[asterisk-users] Auto Reply: asterisk-users Digest, Vol 84, Issue 15
I am out of the office on vacation through July 20th, 2011. I am checking email, and will get back to you as soon as I can. For urgent matters, contact: Angie Besse for Oracle Labs, MA, and Corporate Security Architecture issues. Tami Sisneros regarding Corporate Architecture Approvals. Craig -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Default From and Contact header domain
Hello all, I have a server which is sending INVITEs with a From and Contact header that contains a domain part of the address (an IP address) that I can't explain. My sip.conf does not set a domain. For example in the following line the 123.456.789.012 is the part I can't explain. From: sip:u...@123.456.789.012;tag=aa00104d30 Does anyone know where Asterisk gets the default for these headers from? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Managing the spiralling costs
Wow that's crazy, 1.9 is pretty much as good as your going to get. I would find out where were the most of your traffic is coming from and get local numbers in those areas. When the person calls your 1800 number check if there is a local number for them to use if so play the message with the local number. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vikas Sent: Monday, February 23, 2009 8:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Managing the spiralling costs I have been using the inbound 800 services from vitelity. Slowly the usage has been rising and in the month of Jan the bill was for $650. I am currently on a 1.9 cents a minute plan. Am I paying too much ? Some suggestions my team generated to reduce the toll free incoming call bill were: 1. When people call in on the 800 number take the local number they are calling from and then call them back from our unlimited outgoing account from broadvoice. 2. Find a vendor with a better rate. Any idea what we can do to better manage the 800 cost. Thanks for your time, Vikas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.2/1965 - Release Date: 02/23/09 18:22:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intel Vs AMD
Quad core Intel ;) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson Sent: Sunday, February 22, 2009 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Intel Vs AMD On Sun, 22 Feb 2009, michel freiha wrote: Hi all, I took my decision to use Asterisk server for handling my VOIP calls...My next step is to choose the best hardware that I should use i order to have the best performance...Here I faced 2 choices for my hardware (CPU)... 1- Using Intel CPU or AMD 2- Use 32 or 64 bits Can you help me please to choose between the above choices and what is the advantage and disadvantage of each of choices How many concurrent calls. How much transcoding? A 1GHz Via processor with 128KB cache will handle 100 concurrent calls with no transcoding. Anything above that is a bonus, and 64-bit is a waste for something like asterisk IMO. So use what you're most familiar with. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.2/1965 - Release Date: 02/21/09 15:36:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and PhoneControl
Hi, Has anyone had any experience integrating Asterisk 1.4 with PhoneControl call accounting software ( www.phonecontrol.com.au ) Apparently the s/w does SMDI on serial interface and IP collection. Looking at SMDI in Asterisk I don't think that method will work for SIP calls. Craig ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mitel 5340 IP PHONE
Does anyone have the SIP firmware for a Mitel 5340? Thanks, Craig Van Ham Network Operations PH 1-306-931-8822 Ext: 14 Toll Free: 1-866-328-6144 Ext:14 Email: [EMAIL PROTECTED] Note: The information contained in this e-mail is confidential and may be subject to the rules of privilege. If the reader is not the intended recipient thereof, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify us immediately and delete this e-mail along with any attachments. Thank you. image001.jpg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sonicwall potentially causing long ping times to SIP phones
Can you get another public IP? If so put another router in. Use vlans to seperate the traffic. Sent from my iPhone On 23-Oct-08, at 11:28 AM, James Lamanna [EMAIL PROTECTED] wrote: Bill Michaelson wrote: Sorry for asking the obvious question, but are there other elements of the slow path besides the Sonicwall? I mean, what is in front of the Sonicwall? Also, might the Sonicwall be positioned as some kind of choke point in the topology, thus leading to genuine sporadic congestion? The device in front of the SonicWall is a Cisco Router. Ping times to the ethernet interface of the router are good (~10ms). Also, having a user behind the SonicWall ping the PBX results in an average 20-30ms ping time. So it seems as though the lag is specific to SIP signaling (specifically the OPTIONS requests that asterisk qualify sends out). Unfortunately I can't really ask the client to dump their SonicWall (which we do not manage). On the SonicWall, I know it is configured for Consistent NAT and SIP Transformations are disabled. -- James On Wed, Oct 22, 2008 at 11:35 AM, James Lamanna [EMAIL PROTECTED] wrote: Hi, I'm having an issue where some phones behind a sonicwall are auto- congesting. The status on sip show peer shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify? Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Parking Issue
HI all, I have a question, is call parking broken: When you park a call it says it will time out to a certain extension in a certain context, it never does it just calls the parker back. How do you get it to timeout to certain extension? -- Executing [EMAIL PROTECTED]:2] Park(SIP/testing-b7701418, ) in new stack == Parked SIP/testing-b7701418 on [EMAIL PROTECTED] Will timeout back to extension [craigp] s, 1 in 10 seconds -- SIP/testing-b7701418 Playing 'digits/7' (language 'en') -- SIP/testing-b7701418 Playing 'digits/0' (language 'en') -- SIP/testing-b7701418 Playing 'digits/1' (language 'en') -- Added extension '71' priority 1 to parkedcalls -- Started music on hold, class 'default', on channel 'SIP/testing-b7701418' == Spawn extension (craigp, s, 1) exited KEEPALIVE on 'SIP/testing-b7701418' Thanks, Craig ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sonicwall potentially causing long ping times toSIP phones
I had weird issues when using a Sonicwall, gave up. Stuck in linksys running dd-wrt firmware running on a separate VLAN... no issues since -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Lamanna Sent: Wednesday, October 22, 2008 12:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sonicwall potentially causing long ping times toSIP phones Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on sip show peer shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify? Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] : Parking Issue
HI all, I have a question, is call parking broken: When you park a call it says it will time out to a certain extension in a certain context, it never does it just calls the parker back. How do you get it to timeout to certain extension? -- Executing [EMAIL PROTECTED]:2] Park(SIP/testing-b7701418, ) in new stack == Parked SIP/testing-b7701418 on [EMAIL PROTECTED] Will timeout back to extension [craigp] s, 1 in 10 seconds -- SIP/testing-b7701418 Playing 'digits/7' (language 'en') -- SIP/testing-b7701418 Playing 'digits/0' (language 'en') -- SIP/testing-b7701418 Playing 'digits/1' (language 'en') -- Added extension '71' priority 1 to parkedcalls -- Started music on hold, class 'default', on channel 'SIP/testing-b7701418' == Spawn extension (craigp, s, 1) exited KEEPALIVE on 'SIP/testing-b7701418' Thanks, Craig ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium training course
Fair enough, I did not attend bootcamp, and I passed the dcap at Astricon 2004. My opinion was based on a number of questions in the written exam that I felt had nothing to do with either Asterisk or integration of Asterisk into a customer site. My assumption therefore was that those questions covered content taught in the Bootcamp. I am happy to stand corrected on the matter. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Brentano Sent: Monday, 22 September 2008 1:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium training course I would also disagree that the written exam is biased towards people who attended the training. I attended a Bootcamp earlier this year and thought I was fully prepared to pass the dCAP. Especially since I already had real-world Asterisk experience. But the written exam covered material that we hadn't even discussed in class, some stuff that was in the book, and other that I was totally lost on. I passed the practical with a near perfect score, but fell just short of passing the written. IMHO, the written portion needs to be re-evaluated. What I think needs to change is de-coupling the dCAP exam from the Bootcamp class. I'll likely never retake the dCAP exam since Digium doesn't offer the Bootcamp in my area (Portland) and I can't go to a local testing facility (New Horizons, et al.) and do the exam. It would cost me well beyond the $300 to take the exam after factoring in travel costs and time spent away from work. Also, the problem with the dCAP being coupled to the Bootcamp is that it gives you the false impression that the Bootcamp prepares you to pass the dCAP and that is completely *not true*. In my Bootcamp class of 9 only 4 took the dCAP. Our own instructor said it took him 3 tries to pass! If this isn't going to change, then the dCAP should be changed so that the Bootcamp *does* prepare you to pass. And similarly, Digium should then also offer less expensive (at least, less than $3K) self-study materials or online training that also offers similar training without having to be present at the Bootcamp That way someone could elect to train at their own schedule and later coordinate to drop-in on the last day of a Bootcamp session and take the dCAP. - Chris On Sep 21, 2008, at 9:15 AM, Tilghman Lesher wrote: On Thursday 18 September 2008 20:56:58 Craig Guy wrote: I felt at the time the written portion was heavily biased towards people who had done the training - in fact I would go so far as to say that it was designed specifically to discriminate against people who had not attended the official training. I'd have to disagree with that, having taken the written portion without having attended the bootcamp, and I got one of the highest scores of the people there that day. Included was one question that I believe I was the only that day to have gotten right. Of course, I had the written the application upon which that question was based, so I had an unfair advantage, I suppose. Other than that question, though, I'd have to say that the written portion highly favored the person with a well-rounded set of experiences with Asterisk. However, the test has been revised since I have taken it, and Jared assures me that some of the more tricky questions have been removed, so the written portion may be easier nowadays. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium training course
I got my dCAP by turning up to the exam at Astricon in Madrid a couple years ago without doing any training. It may have changed since then but I found that the practical exam would be difficult if not impossible to pass without knowing what you were doing - either through real world experience or having done the training. I felt at the time the written portion was heavily biased towards people who had done the training - in fact I would go so far as to say that it was designed specifically to discriminate against people who had not attended the official training. Anyhow, the point I am making is that a brain dump will help you pass the written but you'll be humiliated (and rightly so) when you sit the practical. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 19 September 2008 2:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium training course On Thu, Sep 18, 2008 at 12:26 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Tilghman Lesher wrote: On Wednesday 17 September 2008 21:42:20 Steve Totaro wrote: On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno [EMAIL PROTECTED] wrote: Anybody knows how to get a Coupon Code for the discount on the Asterisk training classes??? I am interested on taking that upcoming Asterisk Advance course, and 3K is kinda steep and considering I am still a college student paying this training out of my pocket, every bit helps. Sorry to thread jack. For that matter, I think old timers like myself should automatically get a dCAP. Six or seven years of Asterisk extensive experience should grandfather the dCAP and maybe even the training. I am sure I have a few tricks up my sleeve that the instructors don't know. If memory serves me correctly, there was talk about this very issue when the training and dCAP track came out. I will google it later. Nobody, including Mark Spencer and myself, have gotten a free pass on the dCAP. That said, I think you may be able to take the test for cheap. I don't know the exact price, but in any bootcamp, they allow people to come down on the last day and take only the test, as I did, if they have the space and resources (specifically, for the practical portion). I also used to think the old timers should get grandfathered dCAP cert. Then I looked at some of the stuff the dCAP certification tests for and realized that almost nobody out there that learned Asterisk from the docs and use it in a real world install would be able to pass the dCAP. As Tilghman said you can take JUST the dCAP test for a reasonable fee without having to take the classes. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. Another paper mill to bring down the reputation of the dCAP. Are there braindumps out there, or TroyTec (http://www.troytec.com) 13 page cheat papers that will allow you to hold the highly coveted dCAP? Maybe I should create a site for a nominal donation to the practice tests and braindumps. dCAP is useless if not based on real world experience. That is how I got my CCNA, real world experience. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
I had a look at mine and it has only relays for pins 1,2,4,5 - the other relay positions are on the PCB are not populated. Maybe it has changed recently. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Igor Hernandez Sent: Thursday, 4 September 2008 7:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI Splitter Hy Craig, Can you elaborate on that? In our setup we have it doing just that and it works without a glitch. Regards, Igor H. Craig Guy wrote: The FSV-4PFS as shipped will not switch Ethernet – it switches pins 1,2,4,5. Craig *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *FailSafe Inc. *Sent:* Tuesday, 2 September 2008 11:27 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] PRI Splitter Although the original topic of this thread has changed quite a bit, I wanted to point out that the SPF Product that you are discussing is quite similar to our product, the FSV-4PFS. Ours is a 4 port device which can switch 4 T1/E1/J1/Ethernet or as many as 16 analog lines from a primary to a backup server. It uses similar logic (power outage = failover server, loss of hearbeat = failover server) and also has a physical mechanical switch on the front of it which allows manual override switching to main or secondary server. We also have addressed the 'clean startup' that was discussed a few posts back. The switch will start and remain in 'failover mode' until such time as it receives a hearbeat or the physical switch is moved to the main' position. A failed main server can be restarted/repowered without bothering the backup server operation one bit - until you are ready to switch back to the main server. http://www.failsafevoip.com/index.php?main_page=product_infoproducts_id=1 http://www.failsafevoip.com/index.php?main_page=product_infoproducts_id=1 -- FailSafeVOIP, Inc. Safe is always better than failed http://www.failsafevoip.com [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] On Tue, 2 Sep 2008 00:22:45 +0200, Christian Victor said: that when both servers power fail you have a problem no matter if the failover switch ist still working or not. You've got that right my friend! :-) On Tue, 2 Sep 2008 00:22:45 +0200, Christian Victor said: http://store.variantdistribution.com/category-s/49.htmVariant - one of Rhinos distributors and the only source I was able to find - quotes the card for US$ 700. Strange. I've seen this happen before where retailers will list outrageously high prices for soon-to-be-released products. For example the SNOM KlarVoice handset. MSRP is $32, but I've seen it advertised for $200! http://www.8774e4voip.com/SearchResults.asp?Search=klarvoice I can say with confidence that the LIST price is US $350. The street price will be considerably lower. Frankly, if I were Snom or Rhino I'd be pretty cheezed off about this phenomenon. After hearing the 'buzz' about a new product such as this, I'd hate for customers to *decide* against it mistkenly believing this incorrect price. I'd turn my nose at either of these two products for the incorrect prices I've seen advertised. We're pretty stoked to have stumbled onto this product because it's brand new, and we've been looking for something like it for some time. -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
The FSV-4PFS as shipped will not switch Ethernet - it switches pins 1,2,4,5. Craig From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FailSafe Inc. Sent: Tuesday, 2 September 2008 11:27 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PRI Splitter Although the original topic of this thread has changed quite a bit, I wanted to point out that the SPF Product that you are discussing is quite similar to our product, the FSV-4PFS. Ours is a 4 port device which can switch 4 T1/E1/J1/Ethernet or as many as 16 analog lines from a primary to a backup server. It uses similar logic (power outage = failover server, loss of hearbeat = failover server) and also has a physical mechanical switch on the front of it which allows manual override switching to main or secondary server. We also have addressed the 'clean startup' that was discussed a few posts back. The switch will start and remain in 'failover mode' until such time as it receives a hearbeat or the physical switch is moved to the main' position. A failed main server can be restarted/repowered without bothering the backup server operation one bit - until you are ready to switch back to the main server. http://www.failsafevoip.com/index.php?main_page=product_info http://www.failsafevoip.com/index.php?main_page=product_infoproducts_id=1 products_id=1 -- FailSafeVOIP, Inc. Safe is always better than failed http://www.failsafevoip.com [EMAIL PROTECTED] On Tue, 2 Sep 2008 00:22:45 +0200, Christian Victor said: that when both servers power fail you have a problem no matter if the failover switch ist still working or not. You've got that right my friend! :-) On Tue, 2 Sep 2008 00:22:45 +0200, Christian Victor said: http://store.variantdistribution.com/category-s/49.htmVariant http://store.variantdistribution.com/category-s/49.htmVariant - one of Rhinos distributors and the only source I was able to find - quotes the card for US$ 700. Strange. I've seen this happen before where retailers will list outrageously high prices for soon-to-be-released products. For example the SNOM KlarVoice handset. MSRP is $32, but I've seen it advertised for $200! http://www.8774e4voip.com/SearchResults.asp?Search=klarvoice http://www.8774e4voip.com/SearchResults.asp?Search=klarvoice I can say with confidence that the LIST price is US $350. The street price will be considerably lower. Frankly, if I were Snom or Rhino I'd be pretty cheezed off about this phenomenon. After hearing the 'buzz' about a new product such as this, I'd hate for customers to *decide* against it mistkenly believing this incorrect price. I'd turn my nose at either of these two products for the incorrect prices I've seen advertised. We're pretty stoked to have stumbled onto this product because it's brand new, and we've been looking for something like it for some time. -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PhoneControl integrations
Hi, Does anyone know if it is possible to integrate Asterisk CDR's with PhoneControl software? (www.phonecontrol.com). I think it should be possible, but haven't been able to find any reference to it being done (or even that it can't be done). Craig ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - Nortel CS1K via NRS
Hi, Was wondering if anyone had any tips or experience in getting a Nortel CS1K and Asterisk 1.4.19 to talk to each other via NRS? So far I've gotten asterisk to place calls to the CS1k via the NRS, however calls originated by the CS1K get rejected by the NRS with a 404 Not Found message. If I take the NRS out of the equation by replacing the IP address of the NRS in the CS1K with that of the Asterisk server then everything works ok, however I would like to get the NRS working as it seems to take on the role of SIP proxy server, allowing configuration of multiple SIP trunks where the CS1K seems to be otherwise restricted to a single trunk. Any help appreciated! Craig ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax solution for Asterisk
The Hylafax / Iaxmodem is a good, reliable combination. I have work with a company that competes with eFax using the Hylafax / Iaxmodem combination for termination and also soon for origination. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, 21 May 2008 10:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax solution for Asterisk On Wed, May 21, 2008 at 10:51 AM, Sanjay Rajdev [EMAIL PROTECTED] wrote: We would like to do something similar to efax, where we can send mail to send fax or something similar. I tried to install Asterisk Fax http://asterfax.sourceforge.net/ but was not able to compile it with Asterisk 1.4.19.2, I have read that they recommend Asterisk 1.2.X and older version of SpanDSP. Regards, Sanjay Rajdev Then check out Hylafax and IAXmodem. Hylafax has alot of client apps too. As I said before, it is CPU intensive, so you may need separate machines to handle fax. A direct crossover cable for network is the best to eliminate any latency. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax solution for Asterisk
Not necessarily - if you set your iaxmodems to only produce G4 encoded tiffs you can then use something like c42pdf http://c42pdf.ffii.org/ which essentially copies the tiff image data into a pdf container. Lightning fast, quality is preserved, very little memory usage and very little cpu. I believe that the tiff2pdf binary in later versions of libtiff does a similar thing. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, 22 May 2008 12:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax solution for Asterisk On Wed, May 21, 2008 at 12:00 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Wed, May 21, 2008 at 11:56 AM, Lee Howard [EMAIL PROTECTED] wrote: Steve Totaro wrote: You may need an additional server just to handle faxes if you are running many instances as they are CPU intensive. iaxmodem is not CPU intensive. 100 of them aren't. You can put that many on a typical modern machine and have them all faxing simultaneously and not see a dent in CPU due to iaxmodem. Lee. Hylafax. Iaxmodem doesn't do much good by itself. Thanks, Steve Totaro Probably has more to do with PDFs than tiffs too. I always go with PDF. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Forking in Dialplan
On 4/25/08, Tobias Ahlander [EMAIL PROTECTED] wrote: Date: Thu, 24 Apr 2008 06:54:27 -0700 (PDT) From: Steve Edwards [EMAIL PROTECTED] Subject: Re: [asterisk-users] Forking in Dialplan To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=x-unknown - Tobias Ahlander [EMAIL PROTECTED] escreveu: Is it possible to somehow fork in the dialplan? Say a call comes in. Then I want to wait 30 seconds and then write in a database, but at the same time while I wait I want to go on with other commands too. On Thu, 24 Apr 2008, Vin??cius Fontes wrote: You can call an AGI script that will call another script. That last one would wait 10 seconds and write in the database. The following example works for me: /var/lib/asterisk/agi-bin/agi-test.agi: #!/bin/bash nohup /root/helloworld.sh 1/dev/null 2/dev/null exit 0 /root/helloworld.sh: #!/bin/bash sleep 10 echo Hello world! /root/helloworld.txt exit 0 Why do you need the first AGI? Would: exten = _x.,n,system(nohup /root/helloworld.sh 1/dev/null 21 ) suit your needs? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 Thank you Steve, this seems to work just as I want it to. Now I just have to figure out how to send variables to a system call, but I think I have that covered somewhere :) Best regards, Tobias ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing variables between two DUNDi/IAX2 peers
I believe that IAXVAR in Asterisk 1.6 will do what you want. I have a
backport of this for Asterisk 1.2.14 or so floating around somewhere but it
hasn't been maintained or used for months, may not be compatible with the
1.6 implementation and I offer it with no support whatsoever.
Craig
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vieri
Sent: Thursday, 6 March 2008 2:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Passing variables between two DUNDi/IAX2 peers
--- Richard Lyman [EMAIL PROTECTED] wrote:
Vieri wrote:
Hi.
I am trying to pass a variable from one Asterisk
PBX
to another.
I'm using DUNDi with IAX2. Is there a way to do
it?
I tried the following but it fails.
On peer1:
[dundi-outgoing]
switch = DUNDI/priv
exten = s,1,Set(CDR(userfield)=test)
exten = s,2,Set(DUNDIVAR=${ARG1}#TEST)
exten = s,3,NoOp(Passing ${DUNDIVAR} to DUNDi
peer.)
exten = s,4,Goto(${DUNDIVAR},1)
On peer2:
[dundi-incoming]
exten = _X.,1,NoOp(Received EXTEN ${EXTEN}.)
exten = _X.,n,Set(EXTTODIAL=${CUT(EXTEN|#|1)})
exten = _X.,n,Set(DUNDIVAR=${CUT(EXTEN|#|2)})
exten = _X.,1,NoOp(Extracted extension
${EXTTODIAL}
and DUNDi variable ${DUNDIVAR})
exten =
_X.,n,Goto(local-extensions,${EXTTODIAL},1)
If I try a test call then nothing ever reaches
peer2.
However, if I remove #TEST from DUNDIVAR in
dundi-outgoing and
Goto(local-extensions,${EXTEN},1)
in dundi-incoming then the call is established
correctly.
I guess the _X. pattern match is wrong?
How can I match an alphanumeric string?
Thanks,
Vieri
you would have to use type 'friend' as user/peer do
not pass channel
variables (unless it has been changed in
1.4/1.6/trunk).
In iax.conf I have (on both peers):
[priv]
type=friend
dbsecret=dundi/secret
context=dundi-incoming
and I am running Asterisk 1.2.21.1 on peer1 and
1.2.26.2 on peer2.
Any ideas as to why it's not working?
Or could anyone please suggest an alternative method?
Thanks!
Looking for last minute shopping deals?
Find them fast with Yahoo! Search.
http://tools.search.yahoo.com/newsearch/category.php?category=shopping
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring modem pools in Asterisk [WAS: Connecting a Rolm CBX to Asterisk via T1?]
It should look more like this:
exten = fax,1,Dial(IAX2/iaxmodem1/${NumberCalled}|20)
exten = fax,n,Dial(IAX2/iaxmodem2/${NumberCalled}|20)
exten = fax,n,Dial(IAX2/iaxmodem3/${NumberCalled}|20)
exten = fax,n,Dial(IAX2/iaxmodem4/${NumberCalled}|20)
exten = fax,n,Busy()
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua Kinard
Sent: Wednesday, 27 February 2008 7:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Configuring modem pools in Asterisk [WAS:
Connecting a Rolm CBX to Asterisk via T1?]
Okay, T1 card issue sorted out. New Lesson: Stay Away from TigerJet chips.
Next up, modem pool -- I wanted to know if the below config looked anywhere
near half-sane for defining in asterisk what is essentially a small pool of
four waiting modems that will handle faxes if another modem is busy:
exten = _X.,1,Dial(IAX2/iaxmodem0/${EXTEN})
exten = _X.,2,Busy
exten = _X.,3,Hangup
exten = _X.,4,Dial(IAX2/iaxmodem1/${EXTEN})
exten = _X.,5,Busy
exten = _X.,6,Hangup
exten = _X.,7,Dial(IAX2/iaxmodem2/${EXTEN})
exten = _X.,8,Busy
exten = _X.,9,Hangup
exten = _X.,10,Dial(IAX2/iaxmodem3/${EXTEN})
exten = _X.,11,Busy
exten = _X.,12,Hangup
This seemed logical, but redundant. I've seen the usage of macro's to
condense stuff like that, but I wasn't sure how to have it auto-determine
which modem to use (i.e., iaxmodem0 through iaxmodem3). In my mind, I'm
thinking of this in the form of a for loop:
for each modem in iaxmodem0..iaxmodem3
is it busy?
Yes: Continue
No: Answer
done
done
Is something like that representable in asterisk-speak?
Also pondering ahead for working on outbound faxing, I'm assuming a
[fax-out] context would be somewhat similar as the above, just a different
set of iaxmodems (4-7)?
Thanks!,
--jkinard
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] duplicated voicemail messages
Hello, It has happened to me twice now that duplicated voicemail messages are automatically created, every minute. I have been unable to reliably repeat it (so far), but the basic flow seems to be: 1. a call comes in via my TDM400P (PSTN line) 2. the call is not answered and goes to voicemail 3. the caller does not really leave a message, just 10 seconds or so of silence. At least, that is all I end up with. 4. Every minute from that point on, a new voicemail message is created. All of the messages are 10 seconds of silence, so I assume they are just duplicates of the original message. The first time this happened, my mailbox was completely filled with blank messages. The second time, it just stopped after 25 minutes. In this case I ended up with a CDR indicating that the call was answered and lasted for 25 minutes - although the final destination (dst column) of the call was 't' (which I assume means timeout, not that that makes any sense to me). So, has anybody else ever had a situation where duplicate voicemail messages are created ? And if so, what did you do about it ? Regards, Craig Kowald. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 360 lights not working on subscription
The Linksys SPA962 with SPA932 sidecar support both speed dial and BLF. IMHO very good for the money and very easy to provision once you get a hold of the proper provisioning guide. These things are designed for mass deployment and remote provisioning. As other people have noted, you need to provision via http rather than tftp for best effect. I also have two provisioning files, a shared settings file with the bulk of the config and then a per handset file based on the mac address containing the account and any special customisations. The only bad bit is that a resync usually causes a reboot of the handset which interrupts the connection of anything attached to the PC port of the phone. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: Tuesday, 23 October 2007 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom 360 lights not working on subscription Hey Mike, We started deploying exclusively Polycom and Linksys. The Polycom's support presence, they call it 'Buddy List'. I am not sure about the Linksys phones, I don't think they do although I did see support for SLA (Shared Line Appearance). Omar On 10/23/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: I also have problems with these phones. I have deployed many of them and have had nothing but problems. Omar, what phones did you switch to? I needed some of the features of the snom phones, like the multiple buttons with prescence lights. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar A. Sabek Sent: Monday, October 22, 2007 9:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Snom 360 lights not working on subscription I used to deploy these phones, it was these types of issues that forced me to drop it. It took way too long to troubleshoot the problems and there was a general lack of documentation. This was 2 years ago, things might have changed. If I remember correctly, it was this issue you are having that was the final straw. Good luck, Omar On 10/22/07, Carlos Maimone [EMAIL PROTECTED] wrote: Dear friends, I am working around with a Snom 360 and Asterisk 1.4 + FreePBX In order to get subscriptions working and the Snom 360 lights turns on, I have set everything just like all the pages in the net explain. So, I get subsciption working. I can list subscription on the asterisk and if I use the SIP trace function built in at the SNOM nad see NOTIFY messages and 200 OK responses. But I realized that content length = 0 in all messsages and there isn't any XML content in those Notify headers.. any idea of what's going on? IN SNOM 360 I am currently using firmware 6.5.12 I am pretty sick dealing with this issue. thanks and regards, Charlie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone use the Linksys phones?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Monday, 24 September 2007 6:15 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Anyone use the Linksys phones? Is anyone out there using any of the newer linksys phones since Cisco took over? I am more specifically looking at the spa-941 942's. Just curious about call quality, programability, and functionality with asterisk. I've replaced the Cisco 7960 on my desk with the colour SPA-962. It's about half the buy price of the Cisco, takes less desk space, has more features, and a vastly superior screen. -- I'd like to second the SPA962 - I've deployed a couple of them now and they're great, clients get a kick out of sticking the company logo in colour on the screen and as of fw 5.1.15 the SPA932 sidecar supports asterisk for BLF and speed dial. They're also supposed to to support RSS for stock ticker type scrollies but haven't played with this yet. The only nasty thing I've found is that whenever the handsets resync they reboot even if no settings have changed. When this occurs anything connected to the phones second Ethernet port will drop connection for a few seconds. Craig ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Udev issue on zaptel install
Debian GNU/Linux 3.1 (Sarge). This version supports udev 0.056-3 , but it is not installed as a normal part of the setup process. Which is my problem...probably. Now I have to figure how to set this up. Craig Craig Markham Team Northrop Grumman Arcata Associates Inc. FTTR Lead Instrumentation Engineer W: 775.426.2172 smime.p7s Description: S/MIME cryptographic signature ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Udev issue on zaptel install
Debian GNU/Linux 3.1 (Sarge). This version supports udev 0.056-3 , but it is not installed as a normal part of the setup process. Which is my problem...probably. Now I have to figure how to set this up. Craig smime.p7s Description: S/MIME cryptographic signature ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Udev issue on zaptel install
While attempting to install zaptel I received the following output in response to make install: ... Install -d /etc/udev/rules.d Build_tools /genudevrules /etc/udev/rules.d/zaptel.rules Build_tools /genudevrules :line 1: udevinfo : command not found Make: *** [devices] error 1 And the install aborted. Debian kernel 2.6.17.8-686 Zaptel version 1.4.4 Any ideas? Thanks in advance! Craig smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: DELL Platforms
I've run up to 50 concurrent calls on the PE850 and PE860 using TE205p. I also came across the te110p issue which manifests itself as popping and crackling audio. It is rather insidious as zttest is fine, the problem does not appear to be missed interrupts. In my case the Digium distributor refused to take back the card (we were within the 30 day return period), so I only buy Sangoma now. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joel Hill Sent: Tuesday, 28 August 2007 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: DELL Platforms Hi, About 2 years ago we made the decision to ship exclusively Dell servers. Mostly we have shipped the 860 rackmount with a config of a basic dual core proc couple gig of RAM and a pair of 75GB HDDs in RAID 1. And they are great but we put a limit of about 30 concurrent calls through it. That being said we have got larger installs too, we are running 2 of the older 2950's as a fully redundant load balancing pair. For a call center of around 160. The only thing I would watch for is with the 860 the TE110p doesn't work. The TE120p is fantastic no problems but the older card had some incompatibility. Other than that I've never had one skip a beat, so I hope you have the same luck. Cheers, Joel Hill Support Manager Asterisk IT On Mon, 2007-08-27 at 18:15 -0400, Steve Totaro wrote: Steve Totaro wrote: Arthur Miller wrote: Hello list, I have a customer who is interested in standardizing on dell servers for asterisk deployments. Has anyone had success with a particular configuration? Anything specifically to watch out for? Thank you for your time, Art **Arthur Miller** Sr. Sales Associate **VoIP Supply, LLC**. 454 Sonwil Drive Buffalo, NY 14225 716-250-3871 OFFICE 716-630-1548 FAX [EMAIL PROTECTED] blocked::mailto:[EMAIL PROTECTED] I am running an SC1435 with two dual core Opteron 2212, four gigs of RAM and a couple 250gig SATA drives. Totally VoIP so I cannot comment on cards or interrupts, but so far it has been flawless. I would like to see how many G729/ULAW conversions it could handle. How would I go about benchmarking that? Thanks, Steve Drooling... processor : 0 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 0 siblings: 2 core id : 0 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc processor : 1 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 1 siblings: 2 core id : 0 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc processor : 2 vendor_id : AuthenticAMD cpu family : 15 model : 65 model name : Dual-Core AMD Opteron(tm) Processor 2212 HE stepping: 2 cpu MHz : 2000.000 cache size : 1024 KB physical id : 0 siblings: 2 core id : 1 cpu cores : 2 fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt rdtscp lm 3dnowext 3dnow pni cx16 lahf_lm cmp_legacy svm extapic cr8_legacy bogomips: 4002.32 TLB size: 1024 4K pages clflush size: 64 cache_alignment : 64 address sizes : 40 bits physical, 48 bits virtual power management: ts fid vid ttp tm stc processor : 3 vendor_id : AuthenticAMD cpu family : 15 model : 65
RE: [asterisk-users] basic asterisk knowledge
G729 and annex A differ in the perceived quality and cpu requirements. The annex A version requires less CPU at the cost of loss of quality. The bitstreams are compatible with each other in that a G729A codec can decode a G729 stream and vice versa. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Monday, 11 June 2007 3:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] basic asterisk knowledge On Sun, 10 Jun 2007, Khaled Chehab wrote: I have question concerns asterisk 1-What is difference between G.729 and G.729A? The letter A. http://en.wikipedia.org/wiki/G.729 ... says that G279A uses slightly less CPU to do the compression at the expense of sound quality. Digium appear to supply G279 rather than G729A. (at least they don't mention A) 2-How can I know the requirement hardware for 150 extension on asterisk 1.4.4 making 50 simultaneous call? Google or search the voip-wiki for asterisk scaling, etc. However these days you don't really have much choice - it's a 2.8-3.4GHz Pentium/something or a 1.8-3GHz Xeon/something, or a 3GHz AMD/something. (and their dual/quad processor versions) Basically any modern server class box will do for your needs unless you're transcoding every call. A 3GHz processor and 1GB or RAM will be fine - but you need to be careful with other issues - like making sure disk IO (if doing a lot of call recording/voicemail) won't interfere with Ethernet/Zap/TelcoInterface traffic... I know that 50 simulataneous calls will work fine on a 1GHz processor as long as you're not transcoding. Also, see this: http://www.digium.com/en/products/voice/g729codec.php where they have done some tests themselves and mention the transcoding numbers vs. CPU speed. 3-Do asterisk have a codec conversion? Asterisk will transcode between different codecs, if the codecs are compiled in, or licensed (g729) but transcoding comes at a big CPU cost. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RE: Bottom line on fax reception
I haven't used the iaxmodem / hylafax combo for sending, only for receiving. However my experience is that it is 99% reliable. I am using a Dell PowerEdge 850 with a Pentium 2.8Ghz and 512mb ram. I think it is the Pentium D but could be the dual core, not sure, whatever the base cpu was at the time of order. Running FC4, Asterisk 1.2.16, Hylafax and IAXmodem. Hardware is TE205p with 50 channels active. This combination quite happily receives 50 concurrent faxes without breaking a sweat. Takes roughly 3000 faxes per day. I have another 5 servers similar hardware scattered around the place doing smaller amounts of inbound faxing again with 99% reliability. This same machine also handles inbound voicemail, IVR and converts received tiffs to PDF. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, 29 May 2007 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] RE: Bottom line on fax reception -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jon Pounder Sent: Monday, May 28, 2007 9:10 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] RE: Bottom line on fax reception Quoting Steve Totaro [EMAIL PROTECTED]: If you are a junk spam faxer then it should suit your needs. If you occasionally send faxes and if you do not receive one or the other party does not receive one or it spits out junk but that is OK, then it should fit your needs. If you are faxing contracts or other important documents that are worth something, then go for a more reliable solution. On a 3ghz HP DL320 with a gig of RAM, each fax took about 5% indicated by top. I would not want to go above ten simultaneous faxes so I setup ten IAX Modems (50% in top). Even at that rate, there were a lot of failures. I did not bother to figure out why because these were legal contracts, in bulk, amounting to big dollars. anyone have a comparison with a multicpu machine with the same or lower clock rate ? Let me further qualify my results. This was done with whatever the current stable versions of Asterisk, Hylafax, and IAXmodem were available in January of this year. The faxes were outbound. PDFs put into a Samba share and a cron job moving them over to the Hylafax monitored directory. Thanks, Steve Totaro www.asteriskhelpdesk.com The variables are very simple for any of these kind of decisions. Don't think about savings, think about costs. Costs of equipment Costs of time (resources) implementing Costs of maintenance Costs of losing data (faxes in this case) Costs of going back and doing it the right way if you find the above costs are higher than another solution. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] app_txfax, app_rxfax
That is not true regarding voice / fax detection with iaxmodem. If you are running zaptel, then let it do the fax detection and have the iaxmodems called from the fax context. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Wednesday, 9 May 2007 12:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_txfax, app_rxfax ax. The downside of rx_fax is that you need to compile it into asterisk. The downside of iaxmodem is that (to my knowledge) you can't easilly implement an auto-answer/detect fax/voice/ auto attendant/voicemail system. The channel must be dedicated to faxing, and that's that. This may or may not be an issue for you though. The last fax setup I did was for a small 2-person office where they had an existing fax machine that answered, listened for the remote fax squawk, if it didn't get it, then it rung the phones daisy-chained to it, and if they didn't answer it went to answering machine. I implemented this in asterisk fairly easilly with rx_fax. I'm not sure if you can do that with iaxmodem. Another question along these lines : How does everyone one fax detection on a sip channel? The only thing I've found is NvFaxDetect - anyone know of anything else? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Passing a variable from one Asterisk box toanother
Hi Richard,
there was a thread regarding this a while ago on the dev list which resulted
in a patch being made to allow variable passing via IAX2 channels. See
http://bugs.digium.com/view.php?id=7619 for the patch which I think is in
SVN or anyhow, is not in 1.2
I have recently backported this patch to 1.2 and have a patch which is
tested against 1.2.12, 1.2.12.1 and 1.2.15, but should work against at least
1.2.13 and 1.2.14. The patch introduces a new dialplan function called
IAXVAR, Email me if interested.
Craig
- Original Message -
From: Richard Lyman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 21, 2007 7:27 AM
Subject: Re: [asterisk-users] Passing a variable from one Asterisk box
toanother
Richard Lyman wrote:
Eric Bishop wrote:
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro. All works
great except we now need to pass variables between them.
For example now on box 1 we have:
exten = _23XX,1,SetVar(Foo=1234)
exten = _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
When the call dials into Box 2 the variable Foo does not get passed...
Does anyone have any clever ideas?
as noted in asterisk/docs/README.variables (iirc)
you should see that variable inheritance can occur by prefacing the
variable with '_' or '__'
also, depending on the age of your asterisk you might want to start using
'Set' vice 'SetVar'
also, having ${EXTEN:0} , the :0 doesn't do anything, so you should not
use it and just have ${EXTEN}
i hope this helps
sadly replying to my own post, but, i forgot to mention that
passing variables with IAX2 can be an issue sometimes when you use
user and peer (the user side can pass vars the peer side can not, or
doesn't accept them iirc)
this does not happen using friend, but that has its own issues... check
the wiki for more thoughts about this.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323 to SIP - One way voice
Which H.323 channel driver are you using, and could you post a log or debug of a session. Craig - Original Message - From: Andrei U [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, February 08, 2007 2:41 AM Subject: [asterisk-users] H323 to SIP - One way voice Hello all, I want to use asterisk as protocol converter, H323 to SIP. I am using Asterisk 1.2.14 with chan_h323 and the free version of g729. When calling from SIP to H323 everything is fine. But when calling from H323 to SIP, the phone using SIP doesn't hear the other party. The phones and Asterisk are in the same subnet and the firewall of the Asterisk box is off. Please advice. Thank you, Andrei U ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Asterisk Faxing Support
It's not that Digium don't want fax or t.38 support, it's just that it is not very likely for Steve Underwood to provide it for Asterisk. I'm sure that Digium are very keen for someone to write and contribute t.38 code for Asterisk, it's just that there aren't very many people with the required knowledge and willingness to contribute in that area. The reasons are sorta complex, but as I understand it there are two issues. Spandsp will not be included in Asterisk as Steve will not disclaim the it to Digium, preferring to keep his code under GPL. Likewise, Digium won't accept code that isn't disclaimed - Spandsp could never be included in ABE for example without a disclaimer and it wouldn't make business sense for Digium to have code in the free distribution that can't be in their commercial distribution. The second issue is that it is often very difficult to have code accepted into trunk. An example of this is the t.38 related code that Steve was working on for Asterisk in late 2005. Whilst not directly spandsp, these were backend changes inside asterisk that were required in order to interface t.38 into asterisk. Eventually he gave up and is now focussing his efforts on openpbx which is pure gpl and is easier to get code into trunk, so sort of a path of least resistance - why try to get code into asterisk when it is easier to get it into the fork. Fow now, it is easiest to use hylafax / spandsp with asterisk. The majority of the hard work has been done and Lee Howard is very responsive to user queries. Anyhow, thats my understanding and I could be way off the mark. Craig - Original Message - From: Patrick [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 08, 2007 10:42 PM Subject: Re: [asterisk-users] Re: Asterisk Faxing Support On Thu, 2007-02-08 at 13:55 +0100, Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Asterisk 1.2 has no support of t.38 whatsoever, the call will drop before t.38 is ever utilised, not even pass-thru. 1.4 Adds support for T.38 pass through only and no other sort of faxing, the endpoint must support T.38 and you must send your call to a T.38 gateway and you must not use NAT anywhere in your network and you must enable re-invites which could cause CDRs not to reflect the true details of the call. Asterisk/Digium also has no interest in any further interest in expanding T.38 or faxing support in Asterisk. Steve Underwood and the other fine persons that have helped to develop the software DSPs and other stuff required for FoIP support also have no interest in writing any further faxing support for Asterisk (RxFax, TxFax + the newest span_dsp wont even compile, much less work under Asterisk any more) probably because they know it will never be included into the Asterisk code. Someone please tell me this isn't truth. Afaik it is true that it will not be included in the Asterisk source because Steve will not disclaim the code to Digium (which he off course is entitled to). I compiled the latest spandsp (iirc 0.0.3pre27) on a FC6 box and it compiles fine. On Steve's website there are versions of app_rxfax and app_txfax for 1.4. Takes some messing around with the 1.4 build system to get them included but it worked for me last night. Those apps can be found here: http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/ From reading this list it seems you are better off using iaxmodem and Hylafax (I guess that it assuming the fax comes in via TDM on the Asterisk box). Or check out OpenPBX.org as they have done much work on T.38 support (visit irc channel #openpbx on freenode.net to talk about the current status). Hope this helps. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 problem
Hi Tomislav, It sounds to me that you have t.38 enabled on your Grandstream Handytone 386. You should disable this on the Handytone. I have a handytone 286 which has an option to disable t.38 and use fax passthrough. This should get rid of your t.38 messages on the cli. Craig - Original Message - From: Tomislav Parčina [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 15, 2006 8:40 PM Subject: [asterisk-users] T38 problem I have problem with fax machine Panasonic DX600. It's connected to Grandstream Handy Tone 386 which is connected to Asterisk. Asterisk is connected to my SIP provider. To some numbers I can't send FAX, and I get following error on CLI. WARNING[2237] chan_sip.c: Unknown SDP media type in offer: image 31358 udptl t38 I believe that Panasonic DX600 machine supports T38. And when I have another T38 fax machine on other end they try to send FAX using T38 protocol. And than I believe I get above error and sending FAX fails. Is there any way to solve this? I hear that there is T38 support in Asterisk 1.4, but I can't wait for version 1.4. In manual for Panasonic DX600 I didn't find any instructions how to turn T38 off. Please suggest something. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incorrect Ring tone. Getting a US tone when itshould be AU tone
Hi David, It can be set on the Sipura / Linksys devices. Look under Admin, Advanced, Regional, Call Progress Tones. There is a link floating around on Whirlpool forums to a page and auto provision file containing the correct settings to produce Australian tones. It also depends on whether the phone allows the PBX to make the progress tones or whether the phone alone does them. The setting to control this is in Sipura/Linksys firmware 3.1.10 and higher from memory. If you want the handset to do the tones, or you get a 'double ringing' in the handsets of these phones / ATA's then set Admin / Advanced / Line X / SIP Settings / Sticky 183 to no. Btw, how's your Asterisk going? I'm in the middle of doing a 7 site Least Cost Routed DUNDi setup with redundant routes - Good fun though the learning curve is a bit steep. Craig - Original Message - From: Klaverstyn, David C [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 30, 2006 5:06 PM Subject: RE: [asterisk-users] Incorrect Ring tone. Getting a US tone when itshould be AU tone I don't think it is a phone problem. I get a US ring tone on a PAP2, SPA-942 and IDEFdisk softphone. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Monday, 30 October 2006 5:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incorrect Ring tone. Getting a US tone whenit should be AU tone What phones are you using? It could be a phone level issue. (my aastra has a setting for AU sounds..) PaulH On Mon, 2006-10-30 at 16:13 +1100, Klaverstyn, David C wrote: For some reason Asterisk is producing a US ring tone when it should be an Australian ring tone. I am using ztdummy and do not have any cards installed. My configuration is as follows. I am using Trixbox 1.2.2. Can someone please guide me into the right direction? zaptel.conf loadzone = au defaultzone = au zapata.conf [channels] language=au indications.conf [general] country=au ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Can you explain why multiple registrationisan important (missing) feature ?
I was afraid that may be the case - The issue I have with that approach is
how do you avoid manually mapping extensions to mac addresses in the
dialplan? Assuming I have a PRI with 100did and I want to use the last 4
digits of the DID as the internal extension, I want to use something like
below to handle the bulk of calls:
exten = _,1,Dial(SIP/${EXTEN:4},20)
How can this be accomplished if SIP usernames are mac addresses?, it would
seem to me that sip.conf is the correct place to map an extension to a
device, otherwise I would have an extensions.conf with a manual entry for
each extension making updating it a chore.
Craig
- Original Message -
From: Lacy Moore - Aspendora [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, September 21, 2006 10:23 AM
Subject: Re: [asterisk-users] Re: Can you explain why multiple
registrationisan important (missing) feature ?
On 9/20/06, Craig Guy [EMAIL PROTECTED] wrote:
[9580]
type=peer
auth=000413242fff:[EMAIL PROTECTED]
It would be
[MAC ADDRESS]
type=peer
...etc..
Or at least, that's how I interpreted what Eric said. I think that's an
excellent approach. THe phones are devices. An extension calls one or
more
devices. Makes a lot more sense than multiple extensions calling multiple
extensions.
Your definition in the sip.conf would be defining devices according to
their
MAC addresses. Your dial plan would call these devices based on
extensions.
exten = 100,1,Dial(SIP/MAC) ; where MAC is the MAC address of the phone
--
Lacy Moore
Aspendora, Inc.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature ?
I'm interested, too in how to accomplish this. I have tried earlier today with a Snom360 to register it using its mac address as the authentication username. I can't seem to get it to work (hopefully I'm just doing something wrong). My sip.conf (asterisk 1.2.12) looks something like: [9580] type=peer auth=000413242fff:[EMAIL PROTECTED] With this the handset registers itself with asterisk, however I don't think it is working as I can change the username and password without affecting the registration on the handset. If I try and set secret=secret, or md5secret= then asterisk refuses to register the handset with a 'Registration from ... failed for ... - Username/auth name mismatch' How can I specify the authentication username in sip.conf? Craig - Original Message - From: Tomislav Parčina [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 18, 2006 2:31 PM Subject: [asterisk-users] Re: Can you explain why multiple registration isan important (missing) feature ? In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... And there is your problem. Using the extension as the SIP User ID does not scale, is confusing, and limits your thinking about devices and extensions. There are several reasons this is a bad idea. Multiple extension numbers ringing on the same device / line appearance is the most common. We use the MAC address of the device as the SIP User ID. We append a -a, -b, -c, etc to the MAC address for each line appearance. This does not work well for Softphone, but since All Softphones Suck(TM), we don't really care about this limitation. Users seldom need to know their SIP User ID. Can you please tell me more about this. I don't follow you weary well. I understand that we need to treat phone and users different, but I don't thing that is easy to do with Asterisk 1.2. Maybe something will change, but till then... -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rxfax, spandsp and lack of ecm
spandsp supports 9600 rx and does not support ecm. If you want ecm, use iaxmodem with hylafax - http://iaxmodem.sourceforge.net , currently hylafax in conjunction with iaxmodem seems to be more reliable than rxfax and spandsp by themselves. Craig - Original Message - From: Artifex Maximus [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, September 13, 2006 6:31 PM Subject: [asterisk-users] rxfax, spandsp and lack of ecm Hello, I had received a lot of unreadable pages with rxfax. I've been doing some search on net and found this: http://threebit.net/mail-archive/asterisk-users/msg15708.html It looks like rxfax/spandsp doesn't support ecm error correction. Bad news for me. Is it still the case? app_rxfax.c dated as 8th of february so I think the answer is yes but I am still hoping a little no or might someone have a patch for enabling/implementing ecm. bye, Zsolt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rxfax, spandsp and lack of ecm
It would be nice if someone could do that but I doubt it will happen. Hylafax / iaxmodem is more complicated and more effort to set up than rxfax but the end result is worth the effort. My only criticism is that I set up 2 x E1's on a server (60 channels) and I didn't enjoy having to configure 60 entries in iax.conf, 60 tty's in etc/inittab, 60 modem entries in var/spool/hylafax/etc, 60 entries in extensions.conf .. you get the picture. In that respect rxfax is much much easier and faster to get going, and also more scalable cause you can just keep calling it as many times as you need it without having to know your max concurrent calls in advance. Craig - Original Message - From: Artifex Maximus [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 13, 2006 8:40 PM Subject: Re: [asterisk-users] rxfax, spandsp and lack of ecm Craig Doug, Thanks for your info. I'll do that way. Is there any chance for implementing ecm in rcfax/spandsp? I think using rxfax is more friendly than using a modem emulator connected through a virtual device to a fax software. It's sound as a very bizarre way to me. :-) bye, Zsolt On 9/13/06, Craig Guy [EMAIL PROTECTED] wrote: spandsp supports 9600 rx and does not support ecm. If you want ecm, use iaxmodem with hylafax - http://iaxmodem.sourceforge.net , currently hylafax in conjunction with iaxmodem seems to be more reliable than rxfax and spandsp by themselves. Craig - Original Message - From: Artifex Maximus [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, September 13, 2006 6:31 PM Subject: [asterisk-users] rxfax, spandsp and lack of ecm Hello, I had received a lot of unreadable pages with rxfax. I've been doing some search on net and found this: http://threebit.net/mail-archive/asterisk-users/msg15708.html It looks like rxfax/spandsp doesn't support ecm error correction. Bad news for me. Is it still the case? app_rxfax.c dated as 8th of february so I think the answer is yes but I am still hoping a little no or might someone have a patch for enabling/implementing ecm. bye, Zsolt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rxfax, spandsp and lack of ecm
Try this one: http://www.soft-switch.org/downloads/snapshots/spandsp/ - Original Message - From: Artifex Maximus [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 13, 2006 11:33 PM Subject: Re: [asterisk-users] rxfax, spandsp and lack of ecm Hello Steve, On 9/13/06, Steve Underwood [EMAIL PROTECTED] wrote: Artifex Maximus wrote: Hello, I had received a lot of unreadable pages with rxfax. I've been doing some search on net and found this: http://threebit.net/mail-archive/asterisk-users/msg15708.html It looks like rxfax/spandsp doesn't support ecm error correction. Bad news for me. Is it still the case? app_rxfax.c dated as 8th of february so I think the answer is yes but I am still hoping a little no or might someone have a patch for enabling/implementing ecm. If you look in http://www.soft-switch.org/download/snapshots/snapdsp, the latest snapshot of spandsp and the app_rxfax and app_txfax applications there provide ECM. It is less well tested than the spandsp-0.0.2 code, but seems to be working pretty well now. Sounds promising but gives me Not Found The requested URL /download/snapshots/snapdsp was not found on this server. Apache/2.0.52 (CentOS) Server at www.soft-switch.org Port 80 bye, Zsolt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream Budgetone phones don't show
The lcd in the current budgetone series cannot support alphnumeric display. Craig - Original Message - From: Ricardo Carvalho [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 12, 2006 8:11 PM Subject: Re: [asterisk-users] Grandstream Budgetone phones don't show Thanks Jessee, I've just sent an e-mail to Grandstream support asking if they are planning in a near future to release a firmware implementing alphanumeric callerid for Budgetone series. When they answer me, I'll replay to this thread with their feedback, so the community can also benefit... Regards, Ricardo. Jessee J Holmes wrote: Ricardo, From what I know its a physical limitation of the display Grandstream chose on that phone, Grandstream recommends purchasing the GXP-2000 phone instead if you're looking for this feature. Grandstream has no plans from what I am aware of of making this change to the BudgetTone series phones. You are more than welcome to inquire directly from Grandstream though, this is just from what I know from dealing with them in the past. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Sep 11, 2006, at 11:47 AM, Ricardo Carvalho wrote: I guess this functionality will be in the future added to new firmware releases don't you people think so? Ricardo. Doug Lytle wrote: These phones aren't capable of alphanumeric entries, only numeric. Doug Tom Vile wrote: They only do numeric callerid. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] faxdetect questions - Please HELP!
Hi Bob, in order to stop fax detection, send the call to a context without a 'fax' extension: [incoming] _.,1,doSomeStuff ; Hardfax extension 12345678,1,Goto(hardfax,1000,1) fax,1,receiveFax [hardfax] 1000,1,Dial(Zap/1|70) 1000,n,Hangup - Original Message - From: Bob McDowell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 20, 2006 6:23 AM Subject: [Asterisk-Users] faxdetect questions - Please HELP! I'm using IAXmodem and Hylafax with 'faxdetect=incoming' and things mostly work pretty well. My main lines come in via T1 DID. Today, HR got tired of having someone read and forward their faxes to them and requested we bring their physical machine back on line. I have been able to get the fax forwarded to the appropriate zap channel, but I cannot seem to get it to stop 'faxdetect'ing. After deciding that it is a fax and sending it to the proper zap channel Asterisk says: -- Executing Dial(Zap/5-1, Zap/105) in new stack -- Called 105 -- Zap/105-1 is ringing -- Redirecting Zap/5-1 to fax extension -- Hungup 'Zap/105-1' ...and Hylafax gets it... Now the questions: 1) How can I have 'faxdetect=incoming' for my T1 context and 'faxdetect=no' for my internal zap channels. (I'm assuming that this is what's wrong here...) 2) Is it instead possible to disable faxdetect for the duration of the call? E.g. exten = fax,1,zapFAXDETECT(off) 3) Is there a better way to mix detected faxes and dedicated fax lines? 4) Can anyone share with me a config that accomplishes this feat (both detected and dedicated)? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Receiving faxes and then sending them on
I haven't tried it, but you might be able to do something with the hangup
('h') extension. For example:
[macro-RXFAX]
exten = s,1,Answer()
exten = s,n,Set(FAXFILE=/tmp/fax-${UNIQUEID}.tif)
exten = s,n,Set([EMAIL PROTECTED])
exten = s,n,Set(EMAILADDR=${ARG1})
exten = s,n,rxfax(${FAXFILE}|debug)
[fax-inbound]
exten = _,1,doSomeStuff
exten = _,n,macro(RXFAX,${EMAILADDRESS},${SOMENUMBER})
exten = _,n,Hangup
exten = h,1,goto(dialout,,1)
[dialout]
exten = ,1,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR} $
{CALLERIDNUM})
exten = ,n,Dial(${ARG2})
exten = ,n,txfax(${FAXFILE}|caller)
exten = ,n,Hangup
exten = s,n,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR} $
{CALLERIDNUM})
exten = s,n,Dial(${ARG2})
exten = s,n,txfax(${FAXFILE}|caller)
exten = s,n,Hangup
- Original Message -
From: Koen Van Impe [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, June 16, 2006 8:11 PM
Subject: Re: [Asterisk-Users] Receiving faxes and then sending them on
Maye you should use the 'D' option in the Dial application to proceed when
the call is answered.
Not sure, and I don't have time to test myself, but give it a try!
K
On 6/16/06, Frederik Fix [EMAIL PROTECTED] wrote:
Hi,
I'm trying to setup a system where incoming faxes are received using
SpanDSP and then send on to another (remote) fax machine. The SpanDSP
part is working excellently, however I dont seem to be able to get
the forwarding part to work. Heres what I put into my extensions.conf:
exten = s,4,Answer()
exten = s,5,Set(FAXFILE=/tmp/fax-${UNIQUEID}.tif)
exten = s,6,Set([EMAIL PROTECTED])
exten = s,7,Set(EMAILADDR=${ARG1})
exten = s,8,rxfax(${FAXFILE}|debug)
exten = s,9,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR} $
{CALLERIDNUM})
exten = s,10,Dial(${ARG2})
exten = s,11,txfax(${FAXFILE}|caller)
exten = s,12,Hangup
Asterisk does start dialing at priority 10 however as soon as the
remote fax hangs up that call gets destroyed as well.
Is there anyway to do something like this?
Kind regards,
Frederik Fix
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad BRI card
By the last sentence I mean that only the person or company holding the A-tick can put the sticker on the cards. Paralell importation refers to 'grey' imports that don't come through the vendors sanctioned distribution channels. For example I know that the fritz! has passed approval because this guy has gone through the approval process. The Australian distributor sells them for $400, I can get them off eBay in Europe for $20 per card - the exact same card. $400 is just pure extortion and is going a hell of a long way to prevent the adoption of Asterisk in this country where BRI is the norm and PRI is outrageously expensive. If I had a spare $20k or so then I'd approve the card myself and sell them at a more realistic price. Craig - Original Message - From: Andrew Furey [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 19, 2006 8:54 AM Subject: Re: [Asterisk-Users] Quad BRI card On 5/18/06, Craig Guy [EMAIL PROTECTED] wrote: Any device to legally connect to the PSTN in Australia must be approved by the regulatory body. A process that usually costs at least $20,000 and only allows the permit holder to sell the product for conneciton to the pstn. It is a very high barrier to entry for the Australian market. There is a guy in Victoria who certified the Fritz! card and charges $400 each for them. Paralell imports are not allowed to be connected. Ah, so that's why they're so expensive :( Sorry, what do you mean by that last sentence? Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX over PRI
CPU load definitely affects rxfax - typical symptoms will be pages cut in half. The effect is worse in Asterisk 1.2 than 1.0 - I have a perl agi running and on my test system (PIII 933mhz) the initialisation of the agi takes about 3 seconds. On asterisk 1.2 the exact same agi on the same system takes 12 seconds and pegs the cpu at 100% the whole time. People using festival have also noticed this issue to the extent that festival is unusable on lower end hardware for asterisk 1.2 Craig - Original Message - From: Colin Anderson [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, May 20, 2006 12:48 AM Subject: RE: [Asterisk-Users] FAX over PRI My failure rate, objectively measured, is 3.8%, and this is with 100 - 400 a day. Other than clock slips (which definitely adversely affects fax) I also note that load is an issue. A system with a higher load has a higher probability of failing the fax. Unfortunately, I don't have precise numbers, as I have gotten a feel for this by watching 2 SSH windows to the same box, 1 running top and the other running the Asterisk console. -Original Message- From: Noah Miller [mailto:[EMAIL PROTECTED] Sent: Friday, May 19, 2006 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FAX over PRI Hi Tom - I have had nothing but problems receiving faxes over PRIs with spandsp. I currently have 4 systems, 4 PRIs from 4 different providers... none of them get better than 50% success rates receiving faxes in spandsp, I constantly get cut off pages. No body seems to have a fix for it, and it is really frustrating. Supposedly it is caused by frame slips on the PRI, but if that is the case, I am 4 for 4 getting crappy PRIs that can't keep time. Just an aside thought (sorry to hijack the thread, Steve): 50% - Ouch. I only have one PRI at one of our offices, but we use it to receive faxes that are directly sent via Digium FXS to an analog fax machine. I've never formally tallied up the transmission errors, but we get something close to 100%. Maybe spandsp is an issue here. - Noah On 5/19/06, Tom Christensen [EMAIL PROTECTED] wrote: I have had nothing but problems receiving faxes over PRIs with spandsp. I currently have 4 systems, 4 PRIs from 4 different providers... none of them get better than 50% success rates receiving faxes in spandsp, I constantly get cut off pages. No body seems to have a fix for it, and it is really frustrating. Supposedly it is caused by frame slips on the PRI, but if that is the case, I am 4 for 4 getting crappy PRIs that can't keep time. These same boxes work fine when receiving faxes over fxo ports, or if I plug a fax machine into an fxs port and call in to a spandsp extension the fax will be received just fine, so I am left thinking it must be the PRIs, but if all PRIs are this bad, how can anybody be using them? Tom On 5/19/06, Steve Hanselman [EMAIL PROTECTED] wrote: Sorry for the late reply but both of these are fine, we use spandsp to print some faxes and email others. We also route via a PRI to our other phone system to hylafax on an analog modem and also to an analog fax. So what you want to do is fine and will work. Steve -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Michael Gaudette Sent: 21 March 2006 20:34 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] FAX over PRI Hmmm, Im not so sure I can apply this to me though. I just want to do Fax-To-Email using PRI channels as the incoming lines. Not so much transfer to a real fax. I am assuming that this is easily done with Asterisk? (I did it before with Asterisk SIP, but it only worked once every 10 tries or so) Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: March 21, 2006 3:25 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] FAX over PRI On Tuesday 21 March 2006 15:09, Michael Gaudette wrote: How should I consider Fax over PRI channels with Asterisk? Is the quality and reliability good, or should I be prepared for alot of grief? I'm having good success doing fax over PRI using a TE405; one span to the PRI, the other to an FXS channel bank that is almost obscenely underutilized (3 channels). I also have channel bank - T100P - IAX2 - TE405 - PRI, where the IAX2 link is a 1-hop SDSL (VOIP only) data link. This works well too. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users
Re: [Asterisk-Users] Quad BRI card
From the picture on the web site it looks like it uses a cologne chipset. Any idea if these cards will be available in Australia? Craig - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 18, 2006 9:15 PM Subject: Re: [Asterisk-Users] Quad BRI card stoffell wrote: Aside from being available.. What driver does it use? Will it be needing bristuff ? (that wouldn't work I guess) The Digium B410P will use the mISDN stack and chan_misdn for Asterisk. Or will the near future integrate BRI ( and hfc?) drivers in asterisk? And thus, making bristuff obsolete? (wich means, BRI users will be able to use cvs easily..) No, that will not happen, unless the authors of those drivers want to disclaim them for inclusion into Zaptel and Asterisk. Just to make clear I'm very curious on this card. And yes I'm in europe ;) As another poster mentioned, the B410P card is definitely targeted at the non-US market... not because the card would not work here, but because there is very little availability of BRI lines in the US at all. Most telcos don't even know what they are if you ask :-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad BRI card
Any device to legally connect to the PSTN in Australia must be approved by the regulatory body. A process that usually costs at least $20,000 and only allows the permit holder to sell the product for conneciton to the pstn. It is a very high barrier to entry for the Australian market. There is a guy in Victoria who certified the Fritz! card and charges $400 each for them. Paralell imports are not allowed to be connected. Some manufacturers do the right thing by certifying the card themselves (Eicon for example). Other manufacturers such as AVM leave it to the distributor to certify the card for the local market. The difference is that I can buy an Eicon card off eBay from the US or Europe and legally connect it to the PSTN in Australia as the card comes from the factory carrying the regulartory approval mark. If i was to buy AVM, Digium or Sangoma from another country I'm out of luck cause it doesn't carry the approval sticker that the Australian distributor puts on it. I can understand both points of view - as a customer I want a competitive market so I get value for money. As a distributor I want an exclusive territory so I can jack up the prices to whatever the market will bear without being undercut by nasty competition. Craig - Original Message - From: Michiel van Baak [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, May 18, 2006 10:45 PM Subject: Re: [Asterisk-Users] Quad BRI card On 22:32, Thu 18 May 06, Craig Guy wrote: From the picture on the web site it looks like it uses a cologne chipset. Any idea if these cards will be available in Australia? Can't you just order them from the digium website? Or is digium not shiping to Australia? -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dCAP certification - Advice needed
It sounds like you could pass it. The Boot Camp is not strictly necessary. However I am not sure dCAP would be worth it - I did mine last year and was rewarded with a little plaque saying that I was dCAP certified for asterisk v1.0 - I received the plaque about 1 month before asterisk 1.2 came out. So it seems that you should redo the dCAP every time a new release of asterisk comes out which on the current release cycle means every 6 months. Very expensive investment to keep it current, especially if they don't offer exams in your country and you have to travel internationally to sit it like I did. Craig - Original Message - From: Zach A [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, May 16, 2006 9:01 PM Subject: [Asterisk-Users] dCAP certification - Advice needed Hi dCAP certified people, I want to do dCAP certification and need advice fro you guys. How difficult is it to do it? I am working in asterisk for about 2 years, have installed asterisk systems for a few companies and know quite a bit about asterisk. Is it necessary to go through asterisk boot camp in order to pass the certification test? The boot camp course outline seems very basic and I know much more than that. Please advice me what should I do for the preparation for this certification's test. Zach ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?
If you have both sides of the call it is possible. It may not be practical though. If one side was using spandsp then it is both possible and practical. Craig - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 03, 2006 11:02 PM Subject: RE: [Asterisk-Users] Can I recreate a Fax from a recorded file? Maybe if you had the un-muxed sending side but I really have no idea. Interesting question though. -Original Message- From: Alexander Lopez [mailto:[EMAIL PROTECTED] Sent: Wed 5/3/2006 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: [Asterisk-Users] Can I recreate a Fax from a recorded file? This is a very KGB / NSA / InterPOL / CIA type question, but if I have a recorded file (G.711, no compression) can I feed it into standard in of an application and have it recreate the fax that was send? I don’t know enough about the Fax handshaking to understand this. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreePBX in production?
Wouldn't use it in production for a customer personally. Too many limitations in terms of having a flexible diaplan. What would be nice though is if they were to produce a 'lite' version that gave a gui interface to add/change/move things - sip.conf, voicemail.conf, meetme.conf but staying well away from extensions.conf Craig - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users-List asterisk-users@lists.digium.com Sent: Monday, May 01, 2006 5:19 AM Subject: [Asterisk-Users] FreePBX in production? Has anyone attempted to use FreePBX for a business in production mode? Initial take is there are lots of things scripted but a lot of limitations in terms of supporting basic business functions. Inability (or lack of flexibility) is handling multiple incoming pstn lines, dialplan limitations, poor/no documentation, etc, to mention a few. Maybe its just me, but it appears its no where near usable even with the latest beta1 code. Is it just me or what? Rich ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dreadful results from zttest with TE210P and Dell2850?
Using an SMP kernel will fix the interrupt sharing, you could also disable hyperthreading and set runlevel 3. FWIW I almost exclusively use Poweredge 850 for my * servers with a third party sata raid controller if raid is required. Never had any problems. Craig - Original Message - From: Remco Barende [EMAIL PROTECTED] To: Asterisk Users List asterisk-users@lists.digium.com Sent: Monday, April 24, 2006 6:38 PM Subject: [Asterisk-Users] Dreadful results from zttest with TE210P and Dell2850? Hi list! I'm trying to setup a new * server with a TE210P in a Dell 2850 box. I'm using zaptel version 1.2.5, the linux flavour I'm using is CentOS 4. In a previous thread I read about the results I should expect from zttest. On my home box (using the crappy Asus A7V600) I got really bad results from zttest (just over 97.5) but I know that this motherboard just sucks. To my (huge) disappointment however the results from zttest are equally as bad as from my home box?? (just over 97.5) lspci -vb reveals that the card is sharing IRQ 3 with the second Gbit LAN controller. The box is only idling I'm the only user shh'ing into it. Does anyone have a clue why the results from zttest are this horrible? Looking at the wiki I don't even need to try and put the box into production with such results. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium cards, so disappointing !
I have so far found 2 ATA's that seem to be able to handle FAX reasonably well. The first one is the Grandstream ATA-286 (firmware up to 1.0.6.7, have not tried any firmware later than this), I have used these at multiple customer sites and no one has ever reported problems. They handle G3 faxing ok. Where I work we also use an analog modem connected to one and we get reliable 42k connects. On the asterisk side of things we use PRI (TE110p, TE410p, TE210p). The grtandstreams are plug and go, just disable the t.38 support. The other ATA that I have found to be able to perform faxing is the Linksys PAP2NA, however the configuration is more complex and it doesn't seem to handle G3 or analog modems. I'd recommend the Grandstream as your best chance of successful faxing in an asterisk setting where asterisk has an ISDN connection to the PSTN. Craig - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, April 16, 2006 10:11 AM Subject: Re: [Asterisk-Users] Digium cards, so disappointing ! Remco Barende wrote: Hmm not so sure of that. I have an HP all-in-one thingy. It is not possible to set the TX/RX speed hard in the config at a certain speed. Through the developers menu in the beast it is possible to do this temporary. Faxing at max 9600 bps works, anything higher fails miserably after the second or third page. This doesn't make sense. The known problems are all timing related, and 9600 (I presume you mean V.29 at 9600) is no more or less sensitive to timing slips than V.17. Actually, on a poor line V.17 at 9600bps should perform considerably better than V.29 at 9600bps. Can you tell me your exact setup? There must be something else wrong. I tried lots of different settings but none really seemed to help. The line is ISDN BRI with an HFC-S card. Software is bristuff with florz patch. Echo can, silence suppr. etc all disabled. The HP is connected to a Sipura SPA 2000 with the correct settings for fax and the region i'm in. Still consistently faxes fail after the first or second page. The HP is a LaserJet 3330 mfp. Setting it back 9600 did help a bit. I solved the problem now by connecting an old Digital - Analog converter to the BRI line, bypassing Asterisk. The Sipura is probably the problem. FoIP doesn't generally work for a number of reasons. Packet loss and jitter are just two of them. See http://www.soft-switch.org/foip.html and http://www.soft-switch.org/foip-with-real-atas.html for some others. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk in production as a fax server, anyone?
Hi Adolfo, I have done this and it works. I have maxed out an E1 with 30 concurrent calls of which at least 25 would have been fax. Hardware is nothing special, Dell Poweredge 750, 512mb ram, single SATA drive with either of a TE410p or TE110p card. OS is FC2 with kernel 2.6.9 I expect the server would handle 60 concurrent calls. Asterisk is 1.0.10 with spandsp 0.0.2pre25 and libtiff 3.5.7 Email me privately if you want more details. Craig - Original Message - From: Adolfo R. Brandes [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, March 30, 2006 10:20 PM Subject: [Asterisk-Users] Asterisk in production as a fax server, anyone? Greetings, All-Knowing Asterisk Users List, My company needs to build a reliable fax server that can handle at least 30 simultaneous incoming faxes from the PSTN, using PRI. We realize that this can be solved in any number of ways using a Linux box, but since IVR is also a must, Asterisk popped up as the most promising solution. After combing these lists for clues, we began experimenting extensively with Asterisk and its software DSP and fax capabilities in most of their incarnations, such as Rxfax or Iaxmodem/Hylafax, together with Digium's E1 cards in server-grade Intel motherboards, all in a dedicated test environment. Unfortunately, though, we have yet to achieve reliable and satisfactory results, even with only 1 fax call at a time. I won't go into the details because we don't need technical support, given that this is, as of yet, a very loosely defined test. What we want is is merely a pointer in the right direction. So here it comes: Has anybody ever achieved, or know of someone who has, reliable 30 simultaneous PRI fax calls using Asterisk and Asterisk-compatible hardware and software? We are hardware agnostic, so if you say Sangoma's cards do it better than Digium's, or that Eicon Diva cards' hardware DSP and chan_capi are the only solution, we have no problem going there. I would be most thankful, however, for detailed explanations of successful scenarios, including such things as motherboard make and model, processor speed, Linux distribution and version, and anything else you decide to be even marginally pertinent. Thank you very much, Adolfo R. Brandes ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk in production as a fax server, anyone?
In practice I've found that the fax receiving process is sensitive to CPU load. If the load jumps too high you will see half page fax pages or black streaky pages mixed with perfectly good pages in a multipage fax. Things that can cause this include running agi scripts or rendering your tiff to another format on your * server. I render my faxes on the * server, however received tiffs are queued so as to render them one at a time. If you get page problems you could try rendering them on a dedicated server. Craig - Original Message - From: Lee Howard [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 31, 2006 3:05 AM Subject: Re: [Asterisk-Users] Re: Asterisk in production as a fax server,anyone? Adolfo R. Brandes wrote: Lee Howard wrote: The concurrent calls really isn't that big of a deal, either, if those are your thoughts. The bigger issue seems to be the quality of the audio as it is delivered to the fax application/modem. Interesting. The little information I've found on the subject seemed to imply that Asterisk couldn't handle more than a handful of fax calls using software DSP. This could also be explained away by frame slips too, right? I don't know what a handful means, but fax audio is audio just the same. If Asterisk can handle 30 channels of G.711 then it can handle 30 channels of fax audio. As for the fax application being able to handle it, I know that HylaFAX can handle it; I'm quite certain that iaxmodem could; and I suspect that txfax/rxfax could, but I don't know. The fabled frame slips could account for any number of fax-related problems that users report. Whether or not these things should really be called frame slips is debatable, but I believe that's how the core part of Asterisk sees it - kind of like jitter occurring on a PSTN line. I had begun to get the impression that Sangoma cards were overall better cards than Digium's. It seems that's not necessarily the case. Well, the hardware itself is better, yes, from what I understand. But in my experience that difference doesn't solve the frame slipping issue with the problematic motherboards. The most success I've seen has been to bridge the call through Asterisk to a T1 fax modem such as a Patton 2977 or an Eicon Diva Server with HylaFAX running the modems. Now THAT is a very good idea! To us, it means that if push comes to shove, there is a certain method for having IVR and reliable faxing available during a single call. Thank you! But just to make it clear, woudn't frame slips enter the picture here too? If there is frame slipping on the Asterisk bridge, yes, that could be a problem, too. But in the deployments that I've used there was no so-called frame slipping occurring, so faxing through that bridge was just fine. But, if you're referring to frame slipping occurring on the fax modem... no, I've never [ever] seen that happen with either the Patton 2977 or the Eicon Diva Server (or any of a slew of analog multi-modem hardware)... regardless of motherboard type or sharing of IRQs. I really do tend to believe that the frame slipping problem is with either the Zap hardware or the Zap driver. I don't know how much common hardware there is between the Sangoma and the Digium hardware, but I don't suspect much is common these days, and thus I would tend to look at the zap drivers first as culprits. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk as a fax server
I would recommend using separate servers for inbound and outbound faxing. In my experience outbound faxing is more tricky than inbound (Using spandsp with rxfax/txfax rather than iaxmodem) Craig - Original Message - From: nik600 [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 23, 2006 7:05 PM Subject: [Asterisk-Users] asterisk as a fax server hi is it possible to build a fax server with asterisk? i would like to make a system that: - receives email, converts email and attachments as image and send it via fax - receives fax, converts fax as an image an send it attached in a email to a specific address obviusly the asterisk server is configured with a ISDN card on the phone network... thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe
We went to kernel 2.6.15, and at the same time went to mISDN 0.3.0 rc25, we haven't had any lockups but users are reporting dropped calls. Unfortunately for us this means dropping chan_mISDN in favour of the Cisco router containing BRI cards and then SIP from the Cisco to Asterisk. It may still be possible to use chan_capi with the mISDN drivers for the Drayteks but for us we've run out of time which is a bit of a bummer. I believe the problem is in chan_mISDN which is admittedly still an experimental driver at this stage with release candidates every few days for the past couple weeks. I'm still interested to know how you guys get along with these adapters. As I said, I think the problem is within chan_mISDN at this stage rather than in the USB adapters, so maybe using chan_CAPI on top of mISDN hardware drivers or using chan_vISDN would be the way to go until chan_mISDN matures. Craig - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 13, 2006 3:16 PM Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe Got my 2 dreytek adapters today... Dropped them on to my test system. After wadding thru my Memory of how to setup mISDN, I had it up and running within about 2 hours. You might be receiving an email from me shortly then if I get stuck. If it wasn't for these annoying public holidays (Labour day in Victoria) mine would probably have arrived today too :) Both of them operating in ptmp with no echo cancel turned on at this stage. Seems to be happy. That's quite comforting for initial testing. Could you try some faxing? And is there any way to measure latency with some hard figures, maybe by use of a repeater? Maybe something like this: Echo measurer - BRI 1 - BRI2 - echo responder. Where the measurer dials the responder, sends out a ping, and measures the delay in the response. I find it hard to believe that any USB induced latency could be measurable in milliseconds... Will drop them onto my local production box next week and see how we go :D Let us know! Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe
Hmm, I was using 0.3.0 rc24, or the unstable branch. I see 0.2.0 is listed as 'stable' so maybe I should have used that. Please do keep me informed of your progress. Craig - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 14, 2006 11:46 AM Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of David Phelan Sent: Tuesday, 14 March 2006 13:28 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe Faxing received by SpanDSP seems to work fine with these units. From what I understand, receiving should be more sensitive to delays etc than sending, so it looks like we're onto a winner here! Thanks for reporting back. I'll post with how my testing goes when my unit arrives. james ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe
Being USB 1.1 is not a problem - there is more than enough bandwidth for a BRI in USB. The handsets used in the BRI install are Snom 360's with firmware 5.3 and internal users have complained of slight echo, however I believe this is more to do with the Snoms than the Draytek adapters. For faxing use we have installed a Grandstream ATA 286. I haven't had any feedback yet regarding problems or success with faxing for this customer. I would have expected to hear of any problems faxing by now but I will try to follow it up, however as long as the latency is consistent (ie minimal jitter in the USB stack) it shouldn't cause any problems for fax. At work in our own office we have two SNOM 360's and people with them also complain of slight echo. (We are using TE110p PRI for PSTN). The rest of our office use a combination of Sipura 841, Cisco 7960 and Grandstream BT101 and there are no echo complaints with any of these non Snom handsets, so at this point it doesn't appear that these BRI adapters have echo problems. Craig - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, March 12, 2006 7:35 AM Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe I have ordered one (for $71 from the supplier you mentioned, although I have since found another supplier who appears to have them for $55!!!) and will run whatever testing I can. Someone from Cologne has commented that because it us a USB device, there may be some latency issues (which will amplify any echo problems) and I suspect that faxing may also suffer a bit. They are also only USB1.1, but I'm not sure if that's a problem. Have you tested faxing? Even if faxing doesn't work well enough to be useful because of the delays, I think this is a very nice solution to my problem (lack of BRI hardware in AU). Thanks again for bringing it to my attention!!! James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Craig Guy Sent: Friday, 10 March 2006 14:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe We didn't ask specifically for new ones. I believe the old ones went out of stock a long time ago. We ordered four at once and they all came with the HFC chipset. Craig - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 10, 2006 8:38 AM Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe Note that it is only the currently available minivigors that have the HFCS-USB chipset, older ones on the secondhand market and eBay most likely use a Winbond chipset. Is there any chance that they would sell me an old one? Do I need to ask specifically that they supply the HFC one? Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe
I have been involved with a BRI install using 3 x Draytek minivigor 128 BRI adapters and chan_mISDN. The draytek units use the HFCS-USB chipset, are USB and take power from the USB interface. Each adapter will support PTP, PTMP, TE and I think NT mode with a maximum of 8 adapters (16 channels) per server. The TA's themselves cost $71 inc GST which is the most cost effective BRI / multi BRI solution I have found in Australia to date. I have one in production for about a week, however chan_mISDN is still listed as experimental at this time. Initially with FC4 and the default FC4 kernel the server used to lock up solid about once every 24 hours. It has been suggested to us that people using kernel 2.6.14 or higher do not experience these problems so we rebuilt the server with the new kernel and put it in yesterday. We should find out in a couple of days if this has fixed the lockup problem. If we can't resolve it we'll stick in a Cisco router to handle the BRI. Anyhow, apart from the lockup problem, it does definitely work and if the lockup is in chan_mISDN then you could use chan_capi on top of mISDN with these adapters. I have a server in production elsewhere using the Fritz! card with mISDN drivers and chan_capi for over a year. So, if you have have the ability to do some testing then definitely have a play with these Draytek adapters. I got mine from Netbro in NSW. Note that it is only the currently available minivigors that have the HFCS-USB chipset, older ones on the secondhand market and eBay most likely use a Winbond chipset. As for aesthetics, I was concerned that from the customers viewpoint it might look dodgy, as if we are using the equivalent of external modems to connect the PBX to the pstn, however the units are quite small and have a business feel to them. They look sorta like an ADSL line splitter and cabled neatly look quite professional. Craig - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 09, 2006 6:03 PM Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe Best of luck :-D I would be interested in your progress on this. I am having very little problem in convincing ppl to upgrade their multiple BRI cricuits for a single pri. The cost difference between a te110 (or a Sangoma A101) MORE than covers the difference from the customer stand point, especially once you are up to 3 ISDN-2 Interfaces. A single port E1 is cheaper than any multi BRI adapter I've seen, and based on Telstra pricing, 3.5 BRI services is about the point where the PRI is the cheaper option in terms of monthly rental. Installation cost is another matter but after a year or so it doesn't matter so much. One use for the multi BRI card though, especially one that can do NT mode, is that you can use it to trunk to a legacy BRI PBX, which is why I'm still interested in finding one for use in Australia. James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MultiBRI in Australia - found one - maybe
We didn't ask specifically for new ones. I believe the old ones went out of stock a long time ago. We ordered four at once and they all came with the HFC chipset. Craig - Original Message - From: James Harper [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 10, 2006 8:38 AM Subject: RE: [Asterisk-Users] MultiBRI in Australia - found one - maybe Note that it is only the currently available minivigors that have the HFCS-USB chipset, older ones on the secondhand market and eBay most likely use a Winbond chipset. Is there any chance that they would sell me an old one? Do I need to ask specifically that they supply the HFC one? Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Scripts Terminate too Soon
In the 1.0.x branch asterisk does not always send SIGHUP to agi scripts on call hangup. In 1.2.x a SIGHUP is always sent, even using DEADAGI - From the UPGRADE.txt in the source: AGI: * AGI scripts did not always get SIGHUP at the end, previously. That behavior has been fixed. If you do not want your script to terminate at the end of AGI being called (e.g. on a hangup) then set SIGHUP to be ignored within your application. Craig - Original Message - From: Darren Wiebe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 28, 2006 10:09 AM Subject: Re: [Asterisk-Users] AGI Scripts Terminate too Soon In that case, asterisk sends -HUP to the agi script (I believe). Darren Michael Collins wrote: If that's true, why does dial() return control to the script when the callee hangs up? Doug, if I understand the AGI limitation correctly, the 'dead' in DeadAGI() refers to the other end of a dial() connection. I *think*, but I'm not positive on that. Does anyone know the answer to this one? Thanks, MC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: spandsp 0.0.2pre25
I have used version 0.0.2 every version from pre8 bar pre23 with 1.0.x and pre23, 25 with 1.2.2 and 1.2.4. My libtiff is 3.5.7 with asterisk 1.0.x and libtiff 3.7.1-6 with asterisk 1.2.2 and 1.2.4 I am of the personal opinion through experience that txfax talking to rxfax does not work, and that in any case trying to do more than 3 concurrent txfax is unreliable. I am uncertain of the upper limit of concurrent rxfax, but it is in excess of 12 on TE110p and 1stgen TE4XXp PRI cards. Craig - Original Message - From: Jesse Guardiani [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, February 20, 2006 12:20 PM Subject: [Asterisk-Users] Re: spandsp 0.0.2pre25 Craig Guy cguy at bigpond.net.au writes: Yes, I have compiled and used rxfax succesfully on 1.0.9, 1.0.10 and 1.2.4 to receive from analog fax machines. I have never yet been able to get rxfax working with txfax - my debugs when I try look like the logs in your email. Craig Perhaps I'm just being nitpicky, but you don't mention what version of spandsp you're using. pre20 rtfax - pre20 rxfax works fine here with asterisk 1.0.10 and 1.2.4. I tried using an analog fax machine with pre25 and asterisk 1.2.4 with no luck whatsoever. Unfortunately, I don't have the debug output from those attempts, but I could generate some if it would help. Jesse ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Application Faxing using SIP
I have successfully used the Grandstream ATA286 and Linksys PAP2NA. I would recommend the Grandstream over the Linksys as there is less configuration to do and it is IMHO more reliable for faxes. I have been able to get analog data modem connect at 48k on the grandstream whilst cannot get modem to work at all on Linksys. Craig - Original Message - From: Phil Blundell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 20, 2006 12:10 AM Subject: Re: [Asterisk-Users] Application Faxing using SIP On Sun, 2006-02-19 at 07:36 -0800, Lee Howard wrote: This will work provided that you can create a near-lossless communication path between the ATA and the PSTN gateway (which is the Asterisk box, I assume). One way of creating that, I would expect, would be to add another ethernet card to your Asterisk server and then run a crossover cable between that interface and the ethernet interface of the ATA. You'll also need to configure the ATA to not do lots of things typically done by ATAs, like echo cancellation. That's a good idea. I hadn't thought of using a crossover cable and a dedicated card like that. (Though, that said, I suspect that the datapath through our regular network switches is probably close enough to lossless for this purpose as well.) Any recommendations as to which ATAs are suitable for this purpose? I don't remember seeing a way to disable echo cancellation on either the Grandstream or the Sipura ones that I have here. p. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp 0.0.2pre25
Yes, I have compiled and used rxfax succesfully on 1.0.9, 1.0.10 and 1.2.4 to receive from analog fax machines. I have never yet been able to get rxfax working with txfax - my debugs when I try look like the logs in your email. Craig - Original Message - From: Jesse Guardiani [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, February 20, 2006 1:00 AM Subject: [Asterisk-Users] spandsp 0.0.2pre25 Hello, Is anyone successfully using spandsp 0.0.2pre25 with either asterisk 1.0.x or 1.2.4? I've built a Gentoo ebuild for this version of spandsp and app_rtxfax, and it builds, but I'm not having any luck getting it working. 99% of my test faxes fail. Reverting to 0.0.2pre20 yields a much higher success rate. I've bumped the console debugging level in logger.conf to include debug and verbose, as well as the defaults. I'm running libtiff-3.7.1, because Mr. Underwood's site recommends it, even though it's a vulnerable version of libtiff. I'm faxing from a working, production asterisk 1.0.9 + spandsp 0.0.2pre20 system to a testing asterisk 1.0.10 + spandsp 0.0.2pre25 system, and 1 of 3 things usually happens: 1.) The fax goes through (very rare in testing) 2.) The fax loops indefinitely like this: Feb 19 11:46:04 DEBUG[5089]: chan_zap.c:4059 zt_read: DTMF digit: f on Zap/1-1 Feb 19 11:46:04 DEBUG[5089]: chan_zap.c:4101 zt_read: Fax already handled Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: chan_zap.c:4059 zt_read: DTMF digit: f on Zap/1-1 Feb 19 11:46:07 DEBUG[5089]: chan_zap.c:4101 zt_read: Fax already handled Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089
Re: RE : [Asterisk-Users] Asterisk 1.2.x + oh323 on Debian Sarge.
I'm the coordinator for the OpenH323 project The Ekiga team (previously known as GnomeMeeting) maintain Debian-compatible snapshots of openh323 and pwlib. See the GnomeMeeting (http://www.gnomemeeting.org) download page for more information. Failing that, What versions of openh323/pwlib did you try, and what errors did you get? I'm sure any problems can be fixed, if they have not been already. Craig On Sat, 11 Feb 2006 09:06:19 +0100 Olivier.taylor [EMAIL PROTECTED] wrote: Welcome to the club, same here with freebsd :( -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Jarek Jarzebowski Envoyé : vendredi 10 février 2006 23:01 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Asterisk 1.2.x + oh323 on Debian Sarge. Hello, is anybody there who successfully compiled Asterisk 1.2.4 with oh323 on Debian Sarge? I tried severel versions of oh323 and pwlib and there is no results... only errors. --- Craig Southeren Post Increment VoIP Consulting and Software [EMAIL PROTECTED] www.postincrement.com.au Phone: +61 243654666 ICQ: #86852844 Fax:+61 243673140 MSN: [EMAIL PROTECTED] Mobile: +61 417231046 It takes a man to suffer ignorance and smile. Be yourself, no matter what they say. Sting ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Forwarding out to cellular phone's voicemail with AMP
I have some users who like to forward their extensions out to cellular phones on weekends. They can currently do this using *72cell # which AMP provides. However, in the event that this forward is enabled and a call is forwared to their cell phone but they do not answer it, it will be passed back to Asterisk voicemail. They would prefer the call to continue to ring into their cell phone's voicemail and be left there. How can I go about this? Thanks-- Craig Bruenderman ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: DCAP Certification
As other people have said, the theory exam includes questions not related directly to implementing and supporting asterisk. In addition to knowing asterisk you will need to read up on voip standards. Craig - Original Message - From: Dovid Bender [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 18, 2006 6:21 AM Subject: Re: [Asterisk-Users] OT: DCAP Certification From what I have heard of the training doesnt do to much. The real way to learn is by trial and error. Get the book and start playing. For a while I was trying to use other people's configs have others help me etc. thinking it was the short way out. It ended up being the long way. The best thing is to learn it on your own. Just my $0.02 . Dovid --- Erick Perez [EMAIL PROTECTED] wrote: Hi, emails to astricon.net seems to bounce (at least for me) I need information about proper authorized Asterisk training in the Miami, FL area and the possibility of later DCAP testing. Thanks, -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Spandsp
Yup, do you have /usr/local/lib listed in your /etc/ld.so.conf ? , you may also need to run ldconfig after compiling spandsp, but before compiling rxfax and txfax. Craig - Original Message - From: Tomislav Parcina [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, January 13, 2006 3:57 PM Subject: [Asterisk-Users] Re: Spandsp In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Do you have the spandsp libraries in your library path?, by default they go into /usr/local/lib In that dir I have libspandsp.a, libspandsp.la, libspandsp.so (softlink), libspandsp.so.0 (softlink) and libspandsp.so.0.0.1. Is that all i need to have? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dCAp
I passed the dCap exam at Astricon last year without doing any of the training and it's not easy, it would be very difficult to pass without having practical asterisk knowledge. You really need to know your stuff. However if you have experience with all the things you listed you should be ok. I would suggest you do some background reading on voip history - eg h.323 and mgcp, standards and the asterisk cli. Make sure you know how to configure things like iax.conf, sip.conf, zaptel,conf, zapata.conf, meetme.conf etc etc Craig - Original Message - From: blackgecko [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, January 12, 2006 10:26 PM Subject: [Asterisk-Users] dCAp HI, theres a lot of controversy related to this topic, my company is thinking on me to take the astricon bootcamp, but want to know if it is really whorty, 3000 USD is a huge amount of money to spend, plus the hotel, food and transportation, ive already deployed some asterisk´s pbx and have experience with it using analog tdm cards and E1/T1, queues, conference rooms, IVR, ACD, inbound and outbound routing, VoIP providers like voipjet, etc. Do you think the bootcamp is a good option??? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spandsp
Do you have the spandsp libraries in your library path?, by default they go
into /usr/local/lib
Craig
- Original Message -
From: Tomislav Parcina [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, January 12, 2006 10:32 PM
Subject: [Asterisk-Users] Spandsp
I have tried to install spandsp. On fresh installed FC4 and Asterisk
1.2.1 with zaptel, addons and sounds. I have libtiff, libtiff-devel,
libxml2 and libxml2-devel RPMs installed.
I have untar spandsp-0.0.2pre22.tar.tar and have run
./configure
make
make install
then I have execute patch (at the end of mail) and I didn't recive any
error.
I have again run in /usr/src/asterisk-1.2.1/ dir
make clean; make; make install
and when I tried to start *, it fails when tries to load app_txfax.so.
What could be wrong?
[format_ilbc.so] = (Raw iLBC data)
== Registered file format iLBC, extension(s) ilbc
[app_curl.so] = (Load external URL)
== Registered custom function CURL
== Registered application 'Curl'
[EMAIL PROTECTED] /]#
*** patch file ***
--- Makefile.orig 2006-01-11 18:39:21.0 +0800
+++ Makefile 2006-01-11 18:40:46.0 +0800
@@ -52,10 +52,14 @@
ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/osp/osp.h
$(CROSS_COMPILE_TARGET)/usr/include/osp/osp.h),)
APPS+=app_osplookup.so
endif
+ifneq ($(wildcard $(CROSS_COMPILE_TARGET)/usr/local/include/spandsp.h
$(CROSS_COMPILE_TARGET)/usr/include/spandsp.h),)
+APPS+=app_rxfax.so app_txfax.so
+endif
+
ifeq ($(findstring BSD,${OSARCH}),BSD)
CFLAGS+=-I$(CROSS_COMPILE_TARGET)/usr/local/include -L
$(CROSS_COMPILE_TARGET)/usr/local/lib
endif
CURLLIBS=$(shell $(CROSS_COMPILE_BIN)curl-config --libs)
@@ -100,10 +104,16 @@
rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so
app_curl.so: app_curl.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $ ${CYGSOLIB} $(CURLLIBS)
+app_rxfax.so : app_rxfax.o
+ $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
+
+app_txfax.so : app_txfax.o
+ $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
+
app_sql_postgres.o: app_sql_postgres.c
$(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o
app_sql_postgres.o app_sql_postgres.c
app_sql_postgres.so: app_sql_postgres.o
$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $ ${CYGSOLIB} -
L/usr/local/pgsql/lib -lpq
--
Tomislav Parcina
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: What is the best Dell Machine for Asterisk?
Are you using raid for performance or redundancy? Software raid is nice except when the drive that fails is the one with your boot partition on it. I guess you could always tftp boot the kernel or something. Craig - Original Message - From: Louis-David Mitterrand [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, January 02, 2006 1:17 AM Subject: [Asterisk-Users] Re: What is the best Dell Machine for Asterisk? On Wed, Dec 28, 2005 at 04:02:00PM -0800, William Boehlke wrote: The 830s are nice but limited because they do RAID on a card and have but one suitable PCI slot. So you can have an interface card or RAID, but not both. Linux software raid is, in our experience, much better than any hardware raid solution. We admin 20+ machines all booting on soft raid 1 or 5 partitions up to 2 TB. -- A good friend will help you move, a true friend will help you move a body. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Handytone 486 Outbound problem
I've got my Handytone 486 registered fine with SIP. Using an analog phone attached to it, I can dial 2 digit extensions in the main context just fine. I have a DID mapped to it from the outside which I can also dial and ring through to fine. For some reason thoug, a 7 digit or 10 digit dial string gives back a busy signal to the analog phone.
Asterisk -r makes no mention of any activity when this occurs so it seems that Asterisk is not even generating the busy signal. Is the Handytone capable of doing this and if so, why would it be?I have 20 other Polycom SIP phones configured similarly in the same context which can all dial 2, 7, and 10 digits just fine. They're all just using stdexten Macros.
; 7 digitexten = _NXX,1,Dial(Zap/g1/${EXTEN})exten = _NXX,2,CongestionCould this be a codec problem?-- Craig Bruenderman
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Handyton 486 Outbound problem
I've got a Handytone 486 ATA. It's registering fine with SIP and calls other 2 digit internal extensions just fine. When I try to dial out though (7/10-digit calls), I get a busy signal.How should I troubleshoot this? -- Craig Bruenderman ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Packeteer ? Edgemark ? How to not re-cable ?
I don't recall this being the biz list, but check out www.riverbed.com if you are looking for something that does the job by suppressing repeated traffic rather than compressing or prioritising it. Craig - Original Message - From: Dean Collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 07, 2005 5:21 AM Subject: RE: [Asterisk-Users] Packeteer ? Edgemark ? How to not re-cable ? check out www.exinda.com if you are looking for a cheaper solution to Packeteer, also offers more functionality as the design is third generation. Cheers, Dean From: [EMAIL PROTECTED] on behalf of Stijn Jonker Sent: Tue 12/6/2005 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Packeteer ? Edgemark ? How to not re-cable ? Hello A_Navone, On 06-Dec-2005 21:11, A_ Navone wrote: I have customer wtih 30 stations in cubicles but they only have 1 rj45 per cubicle and that is for lan and internet. I would prefer the voip to be on separate net connection for quality purposes Well I can imagine, or even to protect the softswitch (Asterisk). but customer does not want to recable. How to avoid voice quality problems ? Depending on the usage and switch (not hub) this might even work without seperation. But if you have an switch that supports VLAN's and QoS and your VoIP phone/ata supports VLAN's and QoS supports this, you can run it over one cable. I have read about devices like Edgemark or Packeteer that can prioritize voip udp. Is that true ? Do they work ? A packeteer sounds like some serious $$ to be spend, I'm guessing that recabling might even be cheaper. Why not try the following, there are plugs available that replace the current RJ45 outlets, or plugin an normal outlet and split the cable in 2 times RJ45. See http://www.datorhandel.com/se/products/679-F or http://www.abccables.com/ca-003805.html for an example you would need 1 at both ends, or in the patchroom recable only on that side. This will violate some specifications but depending on cable length and other external influences this might work. In the places I have seen this in place, it generaly works. Generally I'm not a support of this, but if recabling is impossible or to expensive it might be a solution. It's doesn't make the patch rack look any prettier.. ;-) Stijn -- Met Vriendelijke groet/Yours Sincerely Stijn Jonker [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialplan activated Toll restriction
I need to add the facility to allow some of my extensions to be able to dial toll calls by entering a Pin Number to enable toll calling. For example dial *331234567 from any extension to enable Toll calling from extension 123(pin 4567) *34123 from any extension to toll bar extension 123 would prefer it default to toll barred if server restart etc. the code is not really important, the toll bar needs to enable 00xxx dialing. I am sure it has been done before, although I have just spent about 5hours Googling for the answer with no success, could be because I don't have the correct terminology for this functionality. Any help would be appreciated. craig ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linksys SPA-841 Missing Calls
I experienced a similar situation with the SPA-841, it turned out to be that the calls I was missing didn't have caller ID (outside calls with caller ID Blocked), found that the SPA841 phone has an option to ignore calls without caller ID. Turned this option off and it fixed the problem. Sorry, I no longer use the SPA841 and I can't remember the exact menu setting on the SPA841 that fixed it, so you will have to go through the manual. c Message: 1 Date: Fri, 02 Dec 2005 21:43:01 -0800 From: Wolfgang S. Rupprecht [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Linksys SPA-841 Missing Calls To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Might the SPA-841 be crashing and rebooting? With the current firmware (v. 3.1.4) I often see my phone hang and flash all its lights Really? For me the 841 is a quite stable phone. Out of the 15 we have in the office neither one crashed in the past 3 months. And they are used heavily. The phone has weaknesses, but stability in my opinion is not one of them. Phone info: Software Version: 3.1.4(a) Hardware Version: 1.0.0(1813) Elapsed Time: 50 days and 09:48:10 I only have 1 phone so it is hard to tell if the crashing is a hardware or software problem. I never noticed the phone having problems previous to this. I did resync asterisk to HEAD a month ago. Thats also about the time the phone started crashing (or at least I started noticing it). Come to think of it, I've been running the current firmware in the phone since July 20th. The only think that changed in recently was asterisk. I wonder if there is something the newer asterisk is doing that the phone really hates... Asterisk CVS HEAD built by [EMAIL PROTECTED] on a amd64 running OpenBSD on 2005-11-02 00:58:42 UTC Software Version: 3.1.4(a) Hardware Version: 1.0.0(700b) Elapsed Time: 1 day and 05:54:03 (crashed during a call) People have been reporting a finicky ethernet connector, so maybe that is the reason the phone does not answer to any traffic? Yea, this phone has that problem too. ;-) Some cables just don't work. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.10
Can anyone point me to the changelog for 1.0.10? Craig - Original Message - From: Pedro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, November 22, 2005 10:04 PM Subject: [Asterisk-Users] Asterisk 1.0.10 I noticed that asterisk.org http://asterisk.org now has asterisk and zaptel downloads for version 1.0.10 but libpri, addons and sounds are still showing a 1.0.9 version number. Just wondering for those using the 1.0.xversions of asterisk instead of the 1.2 versions - will libpri, addons and sounds be updated to match the 1.0.10version or will 1.0.9 be the final release of those packages? - Pedro ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX difference IAXModem / Hylafax and spandspapp_rxfax
Can Hylafax be made to produce ccitt G4 instead of ccitt G3 encoded images? The G4 tiffs are smaller than G3 and are much more efficient to convert to pdf. I was able to patch spandsp to produce G4 encoded tiffs and was wondering if Hylafax could be made to do the same as I'd really like ECM support. Craig - Original Message - From: Lee Howard [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, November 19, 2005 2:45 AM Subject: Re: [Asterisk-Users] FAX difference IAXModem / Hylafax and spandspapp_rxfax Rob McKrill wrote: I am just getting started in the Hylafax stuff and have a dual PRI card I am trying to do this with, but I am going to give IAXmodem a shot as soon as I get the tif to pdf conversion working. HylaFAX should already have the TIFF - PDF conversion stuff built-in. Actually, it uses libtiff's tiff2pdf, if possible, and if not then uses it's own scripting with tiff2ps - ps2pdf. (The latter being less-good because it uses a Ghostscript tool... which therefore makes it inferior). Lee. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dell and digium hardware
Single port TE110p and quad port TE410p. Craig - Original Message - From: Klaus Darilion [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 17, 2005 1:34 AM Subject: Re: [Asterisk-Users] dell and digium hardware Which digium card do you use? 1 port or 2/4 port E1/T1? or TDM? klaus Craig Guy wrote: I'm using the 850 series. Works well. Only major problem is having to use a third party PCI-e sata raid controller, well thats if you want HW raid in your system. Craig - Original Message - From: Kevin Hanson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 16, 2005 10:08 PM Subject: Re: [Asterisk-Users] dell and digium hardware Klaus Darilion wrote: Hi! I read in the archive a lot of problems using the Dell 1850 servers and digium TE4xxP and TE2xxP hardware. I wonder if ever anybody has tried the Dell Poweredge 850 series and can report some experiences? btw: Does somebody knows why there are problems with 1850 but not with 2850 (digium recommends the 2850 for their Business Edition)? AFAIK both have the same chipset and both use Intel onboard NICs. Thank's for any hints. Regards Klaus We are using a PE 830 at a customer site (single port T1 and TDM10B [for fax machine]). I have had no problems. The only thing I did was use a pci nic and disabled the on-board ethernet. I never tried the on-board, so don't know if it would have caused problems. I just saw a note on Digium's site regarding this (but not for this particular model), and went ahead and got an pci nic. I saw a posting once that indicated that Dell's eighth generation hardware (800's, 1800's, 2800's) don't have problems w/ Digium cards like the 7th gen did (700's, etc). Don't know if this is true, but we went ahead and tried the 830 and have been happy so far. Cheers, Kevin -- Optimacy Communications, LLC http://www.optimacycomm.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 Released!
Any word on when 1.0.10 will be out? I saw mention that 1.0.10 would be released concurrently with 1.2 sometime last week. I've got some issues I am hoping 1.0.10 will help solve. Craig - Original Message - From: Asterisk Development Team [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, November 17, 2005 1:49 PM Subject: [Asterisk-Users] Asterisk 1.2 Released! We are proud to announce that Asterisk 1.2.0 has been released! This release of Asterisk contains over 3,000 improvements on version 1.0, including hundreds of new features and applications. It is available from the ftp.digium.com FTP servers, as well as the Digium CVS servers (under the 'v1-2-0' tag). We want to extend our thanks to all the community members whose contributions have made this release possible; without their coding, support, testing and other involvement we would not have achieved this milestone! Mark Spencer and Kevin P. Fleming (Note: for a short time, a tarball of Asterisk 1.2.0 was present on the FTP servers with a build problem related to the chan_modem drivers; this has been corrected, and if you downloaded the new version before receiving this announcement, please re-download to ensure you have the proper version.) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dell and digium hardware
I'm using the 850 series. Works well. Only major problem is having to use a third party PCI-e sata raid controller, well thats if you want HW raid in your system. Craig - Original Message - From: Kevin Hanson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 16, 2005 10:08 PM Subject: Re: [Asterisk-Users] dell and digium hardware Klaus Darilion wrote: Hi! I read in the archive a lot of problems using the Dell 1850 servers and digium TE4xxP and TE2xxP hardware. I wonder if ever anybody has tried the Dell Poweredge 850 series and can report some experiences? btw: Does somebody knows why there are problems with 1850 but not with 2850 (digium recommends the 2850 for their Business Edition)? AFAIK both have the same chipset and both use Intel onboard NICs. Thank's for any hints. Regards Klaus We are using a PE 830 at a customer site (single port T1 and TDM10B [for fax machine]). I have had no problems. The only thing I did was use a pci nic and disabled the on-board ethernet. I never tried the on-board, so don't know if it would have caused problems. I just saw a note on Digium's site regarding this (but not for this particular model), and went ahead and got an pci nic. I saw a posting once that indicated that Dell's eighth generation hardware (800's, 1800's, 2800's) don't have problems w/ Digium cards like the 7th gen did (700's, etc). Don't know if this is true, but we went ahead and tried the 830 and have been happy so far. Cheers, Kevin -- Optimacy Communications, LLC http://www.optimacycomm.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dell and digium hardware
Works well. I am running 1.0.9 stable on this with FC2 on kernel 2.6.9 The kernel needs patching to pick up the onboard SATA (ICH7), or we use a pci express SATA raid controller with a TE110p. The only real hassle is the single 'standard' pci slot in it. Remote access is via SOL and the embedded third nic. Very nice little server, even cheaper than the equivalent poweredge 750 as we no longer have to buy a drac card. Craig - Original Message - From: Klaus Darilion [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, November 09, 2005 7:06 PM Subject: [Asterisk-Users] dell and digium hardware Hi! I read in the archive a lot of problems using the Dell 1850 servers and digium TE4xxP and TE2xxP hardware. I wonder if ever anybody has tried the Dell Poweredge 850 series and can report some experiences? btw: Does somebody knows why there are problems with 1850 but not with 2850 (digium recommends the 2850 for their Business Edition)? AFAIK both have the same chipset and both use Intel onboard NICs. Thank's for any hints. Regards Klaus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dell and digium hardware
I bought a PCI-e Areca 1210 SATA II raid controller. Who knows what Dell were thinking when they decided to stick a PCI-e slot in the system. http://www.areca.com.tw/products/html/pciE-sata.htm Craig - Original Message - From: Brian Roy [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 09, 2005 9:59 PM Subject: Re: [Asterisk-Users] dell and digium hardware On 11/9/05, Craig Guy [EMAIL PROTECTED] wrote: Works well. I am running 1.0.9 stable on this with FC2 on kernel 2.6.9 The kernel needs patching to pick up the onboard SATA (ICH7), or we use a pci express SATA raid controller with a TE110p. Which pci-e SATA controller are you using? The one that shipped with my dell was pci-x -Brian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wanted to Swap! TDM400 FXO module(s) for FXS
Maybe, but I would expect a fax on a Grandstream ATA-286 would be more reliable than the same fax on the tdm400. I can only speak from my personal experience. I have faxes setup on both the the Grandstream 286 and on linksys PAP2NA, with the ATA's on the same 100mbit switch as Asterisk. The asterisk itself has a PRI connection to the pstn. The grandstream gives the best results - To my knowledge we have never lost a fax on it, and we do dialup modem internet banking with a netcomm modem on the ata286 with reliable 48000bps connect. (The modem will not connect if attached to the PAP2NA). Craig - Original Message - From: Sherwood McGowan [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, October 28, 2005 11:47 AM Subject: RE: [Asterisk-Users] Wanted to Swap! TDM400 FXO module(s) for FXS I agree... I've got wy to many customers out there who are pissed because they thought VOIP would be just as reliable (or even close) as POTS. SKM --Original Message- -From: [EMAIL PROTECTED] -[mailto:[EMAIL PROTECTED] On Behalf Of -[EMAIL PROTECTED] -Sent: Thursday, October 27, 2005 11:27 PM -To: Asterisk Users Mailing List - Non-Commercial Discussion -Subject: Re: [Asterisk-Users] Wanted to Swap! TDM400 FXO -module(s) for FXS - -Agreed. - -PaulH - -- Original Message - -From: Rod Bacon [EMAIL PROTECTED] -To: Asterisk Users Mailing List - Non-Commercial Discussion -asterisk-users@lists.digium.com -Sent: Friday, October 28, 2005 1:12 PM -Subject: Re: [Asterisk-Users] Wanted to Swap! TDM400 FXO -module(s) for FXS - - - Thanks for the suggestion, but in my experience fax -machines on ATAs can -yield - unpredictable results, even at LAN speeds and uncompressed codecs. - - - == - Rod Bacon - Empowered Communications - Ground Floor, 102 York St. South Melbourne - Victoria, Australia. 3205 - Phone: +613 99401600Fax: +613 99401650 - FWD: 512237 ICQ: 5662270 - == - - - Craig Guy wrote: - Consider getting a PAP2-NA to connect your fax machine to -- 2 x FXS - ports for $99 - - Original Message - From: Rod Bacon - [EMAIL PROTECTED] - To: asterisk-users@lists.digium.com - Sent: Wednesday, October 26, 2005 8:46 AM - Subject: [Asterisk-Users] Wanted to Swap! TDM400 FXO -module(s) for FXS - - - Does anyone out there have any TDM400 FXS module(s) that -they want to - swap for FXO (preferably in Australia). - - I have a quad-port FXO arrangement at the moment, but I -need to plug a - couple of fax machines into my * box... - - -- - == - Rod Bacon - Empowered Communications - Ground Floor, 102 York St. South Melbourne - Victoria, Australia. 3205 - Phone: +613 99401600Fax: +613 99401650 - FWD: 512237 ICQ: 5662270 - == - ___ - --Bandwidth and Colocation sponsored by Easynews.com -- - - Asterisk-Users mailing list - Asterisk-Users@lists.digium.com - http://lists.digium.com/mailman/listinfo/asterisk-users - To UNSUBSCRIBE or update options visit: -http://lists.digium.com/mailman/listinfo/asterisk-users - - - ___ - --Bandwidth and Colocation sponsored by Easynews.com -- - - Asterisk-Users mailing list - Asterisk-Users@lists.digium.com - http://lists.digium.com/mailman/listinfo/asterisk-users - To UNSUBSCRIBE or update options visit: -http://lists.digium.com/mailman/listinfo/asterisk-users - - - ___ - --Bandwidth and Colocation sponsored by Easynews.com -- - - Asterisk-Users mailing list - Asterisk-Users@lists.digium.com - http://lists.digium.com/mailman/listinfo/asterisk-users - To UNSUBSCRIBE or update options visit: -http://lists.digium.com/mailman/listinfo/asterisk-users - - -___ ---Bandwidth and Colocation sponsored by Easynews.com -- - -Asterisk-Users mailing list -Asterisk-Users@lists.digium.com -http://lists.digium.com/mailman/listinfo/asterisk-users -To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk
Re: [Asterisk-Users] Astricon - materials
Any word on the availability of the Madrid materials? Craig - Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, October 26, 2005 2:42 PM Subject: Re: [Asterisk-Users] Astricon - materials marek cervenka wrote: hi, will be somewhere materials (videos, presentations) from astricon? Registered attendees will get information about the material soon. No videos where recorded this year. The 1.2 presentation I made together with Kevin has been available for a while at http://www.astricon.net/asterisk1-2/ and will be updated soon. Regards /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wanted to Swap! TDM400 FXO module(s) for FXS
Consider getting a PAP2-NA to connect your fax machine to - 2 x FXS ports for $99 - Original Message - From: Rod Bacon [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, October 26, 2005 8:46 AM Subject: [Asterisk-Users] Wanted to Swap! TDM400 FXO module(s) for FXS Does anyone out there have any TDM400 FXS module(s) that they want to swap for FXO (preferably in Australia). I have a quad-port FXO arrangement at the moment, but I need to plug a couple of fax machines into my * box... -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk
Do you have a permit line in manager.conf for connections from 127.0.0.1 such as: permit = 127.0.0.0/255.0.0.0 And also a bind entry: bindaddr = 0.0.0.0 Craig - Original Message - From: Chuck Bunn [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 17, 2005 1:21 PM Subject: Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk Hi, Yes it is enabled I have even checked various logs and nothing... I checked '/var/log/messages', '/var/log/secure', '/var/log/asterisk/full', and even '/var/log/mysqld.log' nothing, nada, nein - its odd that a failed connection attempt is not logged somewhere, perhaps I must somehow turn logging on for the asterisk management portal. Any ideas? Thanks [EMAIL PROTECTED] wrote: On 10/17/2005, Michael Furdyk [EMAIL PROTECTED] wrote: He is just using telnet to check for the port being open/working... (not telneting to the telnet port) -- Mike -Original Message- [EMAIL PROTECTED] Sent: Monday, October 17, 2005 12:28 AM Subject: Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk On 10/17/2005, Chuck Bunn [EMAIL PROTECTED] wrote: Hi, I cannot do the following: telnet 127.0.0.1 5038 Is telnet enabled? Brett Here it is Sunday - And I been wrong already this week... Is manager.conf 'enabled=yes'? Brett ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with compiling spandsp
Download the latest app_rxfax.c and app_txfax.c for pre21 (Dated 12 October 2005). For the first week or so pre21 was available the older versions were posted by mistake and caused exactly this compilation error. Craig - Original Message - From: Administrator [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 18, 2005 3:53 AM Subject: RE: [Asterisk-Users] Problem with compiling spandsp Actually I am using 0.0.2pre21, also tried pre20finally got a different error after trying just about everything including deleting the source dir and unpacking again, editing makefile again, etc. app_rxfax.c: In function `rxfax_exec': app_rxfax.c:265: error: structure has no member named `logging' app_rxfax.c: At top level: app_rxfax.c:61: warning: 't30_flush' defined but not used make[1]: *** [app_rxfax.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.0.9/apps' make: *** [subdirs] Error 1 Maybe I'm not editing the makefile correctly? I am cutting/pasting from the patchfile so I know it's not a typo. -Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Friday, October 14, 2005 6:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problem with compiling spandsp Administrator wrote: New asterisk user, pretty much set up except for spandsp. I get the following when trying to compile: app_rxfax.c app_rxfax.c: In function `phase_e_handler': app_rxfax.c:92: error: structure has no member named `cid' app_rxfax.c:92: error: structure has no member named `cid' app_rxfax.c: In function `rxfax_exec': app_rxfax.c:260: error: structure has no member named `verbose' app_rxfax.c: At top level: app_rxfax.c:61: warning: 't30_flush' defined but not used make[1]: *** [app_rxfax.o] Error 1 I'm running and compiling against Asterisk 1.0.9 on a CentOS4_x86_64 system. Asterisk alone compiles and is running without issue. I can't find any problem with dependencies. Any help would be appreciated. I had the same issues with .0.0.3 and went back to the 0.0.2 version 0.0.3 is for developers. Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?
I have downloaded iaxmodem and gone through the readme but not yet installed it. I currently use rxfax to receive in the vicinity of 1200 faxes per day and 5000 or more pages (faxes vary from single page to 30 pages) per E1, with a peak load of about 12 concurrent inbound faxes to rxfax. Best I can tell my failure rate is about 0.8%. I have been testing using Hylafax for faxout with an 8 port analog fax modem card and a couple PAP2NA's and this works well, but I am very much looking forward to checking out iaxmodem. Especially if using Hylafax will give me ECM. Craig - Original Message - From: Lee Howard [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 13, 2005 10:47 AM Subject: Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable? Darren Nickerson wrote: We prefer the Eicon Diva server and Brooktrout TR1034 boards, which are known to work well with HylaFAX since we've had our share of headaches with the 2977's. Well, part of my preference for the 2977s involves my strong dislike for the way that the Diva Servers and BrookTrouts do things. It's enough of a dislike to get me over the learning curve of how to properly set up the 2977s for HylaFAX use. Having said that, I'm excited to see Lee and Steve improving IAXmodem and the underlying SpanDSP library, and look forward to the day that is performs similarly (or better) to the DSP-laden boards we presently favor! If your favor involves V.34 then it may be a while before the relevant patents expire. Lee. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable?
I'm trying to figure out what an appropriate deployment model might be. Whether to have iaxmodem installed on the hylafax server with a switched ethernet connection for iax2 to the * server with the PRI, or to have the iaxmodem on the PRI * server and channel the tty comms across the network. I suspect that the latter might work ok over a WAN so I could have a central hylafax server with distributed * servers running iaxmodem at the far end of wan links (up to 100ms latency). The only issue is that I want to retain rxfax on the PRI * servers for incoming faxes. Lee, if I install iaxmodem on a * server for outbound faxing from hylafax, can I still use rxfax on the same server to receive faxes? Craig - Original Message - From: trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 13, 2005 3:06 PM Subject: Re: [Asterisk-Users] Which asterisk-friendly cards are fax-capable? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soekris and Asterisk
It will if I stick the 4801 in a bigger case :) Craig - Original Message - From: BJ Weschke [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 11, 2005 12:40 PM Subject: Re: [Asterisk-Users] Soekris and Asterisk The quadspan card isn't a low profile card is it? I don't think it'll even physically fit in the net4801's footprint. On 10/11/05, Craig Guy [EMAIL PROTECTED] wrote: Has anyone on the list used a Soekris engineering PC as a TDM - Ethernet bridge? For example something like a net4801 with a TE110p in it and then using TDMoE to get it into a bigger server where the call processing proper will occur. Anyone know if it might handle a quadspan card ok? (no transcoding, just pure PRI to TDMoE bridging). Craig ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com-- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soekris and Asterisk
Hi Kristan, The interrupt load is what I am most worried about. At most it would be two spans. Echo cancellation is not required as 80% or more would be fax traffic, the rest IVR and voicemail. I am aware of AstLinux but unfortunately for this particular application the Soekris OS is gonna be FreeBSD as the Soekris is primarily a router with the PRI piggybacking. As far as I can tell, I don't need asterisk installed, just zaptel and libpri. I guess I'll find out. The major reason for this is that I can't physically stick the PRI card in my * server (don't ask!) so this is one of the alternatives I have dreamed up, along with setting up the Soekris with the PRI and IAX2 trunking to the * box but I think TDMoE would be much more efficient. At the end of the day I might just have to get a bigger * server to get the card in, but doing it this way would be an interesting hack and also allows some sort of scripted failover. Eg, if the Soekris sees that a * server has died it can stop the zaptel service, swap in different config files pointing to a mac address in a backup * server and away we go :) Craig - Original Message - From: Kristian Kielhofner [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 11, 2005 6:35 PM Subject: Re: [Asterisk-Users] Soekris and Asterisk Craig Guy wrote: Has anyone on the list used a Soekris engineering PC as a TDM - Ethernet bridge? For example something like a net4801 with a TE110p in it and then using TDMoE to get it into a bigger server where the call processing proper will occur. Anyone know if it might handle a quadspan card ok? (no transcoding, just pure PRI to TDMoE bridging). Craig Craig, It all depends on where you are going to do what (PRI, echo cancel, etc). Also, for four spans the interrupt load alone could probably saturate the CPU. If you want to try, AstLinux will be an excellent start... http://www.astlinux.org P.S. - I created AstLinux, so of course I would recommend it! -- Kristian Kielhofner ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soekris and Asterisk
Cool, now if only it was available in E-1, and certified for use in Australia. Actually this is pretty much what I was thinking of building myself :) Now I know it can be done. Yippee! Craig - Original Message - From: astgroups [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 11, 2005 9:18 PM Subject: Re: [Asterisk-Users] Soekris and Asterisk You should look at the Redfone fonebridge product. I believe their product does what you are wanting to do; http://www.red-fone.com/fonebridge.html On Tue, 2005-10-11 at 00:38, Craig Guy wrote: Has anyone on the list used a Soekris engineering PC as a TDM - Ethernet bridge? For example something like a net4801 with a TE110p in it and then using TDMoE to get it into a bigger server where the call processing proper will occur. Anyone know if it might handle a quadspan card ok? (no transcoding, just pure PRI to TDMoE bridging). Craig ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: www.openpbx.org
- Original Message - From: asterisk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 11, 2005 12:15 AM Subject: Re: [Asterisk-Users] Re: www.openpbx.org The other thing that I think many are missing is the recent deal with Intel and finally I remember that the Digium backed Asterisk Certification was unfair and pricy since many guru developers would still need to take the exam to become certified just to line a few people's pocket even thought they probably know more than the people teaching the cert course. I don't really want to get sucked into the whole openpbx thing but I did just want to comment one point in this part: I took the opportunity to do the Asterisk Certification Exam at Astricon Europe (I did not do the training course, however I did manage to pass). My impression of the multi choice 'theory' part of the exam is that it was written deliberately to encourage people to undertake the paid training course. A number of the questions were involved with stuff that someone building asterisk systems would never ever have to deal with or think about such as the vendors behind some of the VOIP standards, other esoteric historical information that would never be used, and various obscure asterisk command line switches and cli commands. Of course, I'm sure that the paid training course has a couple hours devoted to such things. The practical part of the exam showed a distinct USA bias - It was in terms of T1's and analog zap extensions. I am from Australia, and the exam was in Europe, these parts of the world generally use BRI ISDN and PRI E1 with hdb3 and crc4 line protocols and channel 16 as the D channel. I'm not sure about Europe, but in Australia up until very recently the Zaptel analog cards were not certified for connection to the PSTN, which makes knowledge of them irrelevant for this part of the world. I don't know how to configure a T1 and I probably will never need to in my * career. The certification testing should be regionalised for the specific country or part of the world it is being administered in. Since the exam I have heard nothing, no congratulatory email, no certificate with a dCAP membership number, no login to a website or dCAP community forum etc. No access to digium or asterisk logos to put on my business cards or website, no listing of certified people on the Digium website. So at the moment I don't really see what benefit there is to paying a couple hundred dollars for the exam. Sure, I tell people that I am certified, but if they ask for proof I have none to give. I did email Digium about this and received a vague reply about printing up and mailing out some plaques at some time in the future. To me it almost seems like Digium are treating their dCAPS as competition rather than partners given the lack of support to date. Craig Thanks, Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AVM Fritz! + chan_capi + mISDN + PTP
Hi,
Yes, you can use the Fritz! in PTP mode, though only if you are using the
mISDN drivers. The mISDN driver should be called like this:
modprobe avmfritz protocol=34
Craig
- Original Message -
From: Lionel Riem [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, October 10, 2005 4:04 PM
Subject: [Asterisk-Users] AVM Fritz! + chan_capi + mISDN + PTP
Hello everyone,
I have been using an AVM Fritz! card with chan_capi and mISDN for quite a
while in PTM mode and it was working finely.
Now, I needed more DID/MSN, so I switched to PTP. But now nothing works
anymore :(
I am using Asterisk on Debian Sarge stable and installed Asterisk along
with chan_capi from apt-get. I installed mISDN from the CVS of
isdn4linux.de.
It is :
- Asterisk 1.0.7 with bristuff
- chan_capi 0.3.5
When I load the whole modules lot, I get the following in dmesg:
Modular ISDN Stack core $Revision: 1.25 $
mISDNd: kernel daemon started
ISAC module $Revision: 1.16 $
mISDNd: test event done
CAPI Subsystem Rev 1.1.2.8
capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs)
ISDN L1 driver version 1.11
ISDN L2 driver version 1.20
mISDN: DSS1 Rev. 1.30
mISDN Capi 2.0 driver file version 1.14
X25 DTE modul version 1.8
AVM Fritz PCI/PnP driver Rev. 1.30
ACPI: PCI interrupt :00:14.0[A] - GSI 10 (level, low) - IRQ 10
mISDN_fcpcipnp: found adapter Fritz!Card PCI v2 at :00:14.0
fritz card cd09a000 dch cd09a094 bch1 cd09a214 bch2 cd09a3a0
AVM PCI V2: stat 0x240020e
AVM PCI V2: Class E Rev 2
AVM PnP: HDLC version 2
mISDN: AVM Fritz!PCIv2 config irq:10 base:0xEC00
spin_lock_adr=cd09a024 now(d015b867)
busy_lock_adr=cd09a024 now(d015b867)
AVM PCI/PnP: reset
AVM PCI/PnP: S0/S1 40/2
Fritz1 ISAC STAR 40
Fritz1 ISAC MODE c0
Fritz1 ISAC ADF2 ff
Fritz1 ISAC ISTA 0
Fritz1 ISAC CIR0 7
mISDN_isac_init: ISACSX
Fritz1 HDLC 1 STA 8200
Fritz1 HDLC 2 STA 8200
AVM Fritz!PCI: IRQ 10 count 4
fritz 1 cards installed
Here is my /etc/asterisk/capi.conf:
;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
mode=immediate
isdnmode=ptp
msn=*
incomingmsn=*
controller=1
softdtmf=1
context=dispatcher
accountcode=
devices=2
Here is my /etc/modprobe.d/capi conf file:
alias /dev/capi20 avmfritz
alias char-major-68-0 avmfritz
install avmfritz /sbin/modprobe capi; \
/sbin/modprobe mISDN_core; \
/sbin/modprobe mISDN_l1; \
/sbin/modprobe mISDN_l2; \
/sbin/modprobe l3udss1; \
/sbin/modprobe mISDN_capi; \
/sbin/modprobe mISDN_x25dte; \
/sbin/modprobe --ignore-install avmfritz protocol=0x22
remove avmfritz /sbin/modprobe -r --ignore-remove avmfritz; \
/sbin/modprobe -r mISDN_x25dte; \
/sbin/modprobe -r mISDN_capi; \
/sbin/modprobe -r l3udss1; \
/sbin/modprobe -r mISDN_l2; \
/sbin/modprobe -r mISDN_l1; \
/sbin/modprobe -r mISDN_core; \
/sbin/modprobe -r capi
capiinfo shows me:
asterisk:/etc/asterisk# capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: mISDN CAPI controller Fritz1
CAPI Version: 2.0
Manufacturer Version: 1.0
Serial Number: 0002
BChannels: 2
Global Options: 0x0018
DTMF supported
Supplementary Services supported
B1 protocols support: 0x0003
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
B2 protocols support: 0x0043
ISO 7776 (X.75 SLP)
Transparent
Transparent (ignoring framing errors of B1 protocol)
B3 protocols support: 0x0005
Transparent
ISO 8208 (X.25 DTE-DTE)
0100
0200
1800
0300
4300
0500
Supplementary services support: 0x0012
Terminal Portability
Call Forwarding
In Asterisk, when an incoming call arrives, it shows me the following:
Asterisk Ready.
*CLI capi info
Contr1: 2 B channels total, 2 B channels free.
*CLI capi debug
CAPI Debugging Enabled
*CLI
*CLI
*CLI -- INFO_IND ID=001 #0x0001 LEN=0016
Controller/PLCI/NCCI= 0x101
InfoNumber = 0x18
InfoElement = 89
-- INFO_IND ID=001 #0x0001 LEN=0016
Controller/PLCI/NCCI= 0x101
InfoNumber = 0x18
InfoElement = 89
Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1184 find_pipe: PLCI doesnt
match last pipe (PLCI = 0x101)
Oct 10 09:17:16 NOTICE[5948]: chan_capi.c:1301 pipe_msg: INFO_IND ID=001
#0x0001 LEN=0016
Controller/PLCI/NCCI= 0x101
InfoNumber = 0x18
InfoElement = 89
-- CONNECT_IND ID=001 #0x0002 LEN=0044
Controller/PLCI/NCCI= 0x101
CIPValue= 0x1
CalledPartyNumber = 8120
CallingPartyNumber = 01 830123456789
CalledPartySubaddress = default
CallingPartySubaddress = default
BC = 80 90 a3
LLC = default
HLC
[Asterisk-Users] Soekris and Asterisk
Has anyone on the list used a Soekris engineering PC as a TDM - Ethernet bridge? For example something like a net4801 with a TE110p in it and then using TDMoE to get it into a bigger server where the call processing proper will occur. Anyone know if it might handle a quadspan card ok? (no transcoding, just pure PRI to TDMoE bridging). Craig ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp and page orientation
Hi Shawn, Could you explain what you mean by 'orientation'. Are your faxes rotated 90 degrees?, are they compressed in the longitudinal plane? Send me one of your landscaped tiff files offlist and I'll try to see whart is going on. Craig - Original Message - From: Shawn Porter [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, October 04, 2005 10:31 PM Subject: [Asterisk-Users] spandsp and page orientation I have just installed spandsp-0.0.2 onto my Asterisk 1.0.9 I am using an old Intel 536EP (actually found drivers that work) BUT...all my faxes are coming in landscape mode Has anyone come across this? any fixes? Shawn ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T.38 Faxing - at astricon ?
The problem as I see it is that if people start expecting it to work then rather than being pleasantly surprised when it does, they will be bitterly disappointed when it doesn't. IMHO analog fax over IP is too flaky to encourage the general public to utilise, and any suggestion to the contrary is misleading. Having said that, I have an analog fax connected to an ATA that works 100% of the time, however I have my ATA and Asterisk on the same ethernet switch. I wouldn't expect to have it work reliably over a WAN or other broadband internet connection. Craig - Original Message - From: Roger Schreiter [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, September 29, 2005 10:31 PM Subject: Re: [Asterisk-Users] T.38 Faxing - at astricon ? Roy Sigurd Karlsbakk schrieb: ... see http://soft-switch.org/foip.html for a brief explaination of why this generally doesn't work... Hi, maybe one should update this link. I think, you agree, that VoIP is somewhat similar to ISDN, as it transports analog audio data in a digitally coded way. Noone doubts, that ISDN is suitable to transport analog fax. Finally the PSTN is 99,9% digital (ISDN/SS7), even if some subscriber lines are still analog. (Ok, ISDN is a managed network, and thus very high quality.) Since there are more and more regions in the world, where internet connectivity quality approaches to ISDN quality, analog faxing over VoIP becomes reliable and hassle free. You should have 128kbit in both directions, better 256kbit, maybe some QoS build in your router (e.g. Linux's iproute2), and pingtimes below 20ms to the VoIP-provider (PSTN-gateway). DSL with fastpath or internet by TV cable does provide this standard imho and become more and more available. Thus we shouldn't discourage people generally of faxing, even if there are a lot of trouble reports. Who can count the success stories with (analog) fax over IP, which are not posted? As far as I see, there are more users faxing without observing quality differences to ISDN than users with problems with fax over VoIP. This is, what various partners of ours do report after having replaced BRI connections by VoIP in some small and middle sized companies. Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for firmware for Cisco 12sp+ and 30VIP
I have been looking for the firmware for the 12sp+ and 30VIP and have been unable to find it. Any help in locating would be much appreciated. Thanks===This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the sender immediately. This message may contain confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
